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  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Swiss Parliaments Corpus, an Automatically Aligned Swiss German Speech to Standard German Text Corpus</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Michel Pl u¨ss</string-name>
          <email>michel.pluess@fhnw.ch</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>Lukas Neukom Christian Scheller Institute for Data Science University of Applied Sciences and Arts Northwestern Switzerland Windisch</institution>
          ,
          <country country="CH">Switzerland</country>
        </aff>
      </contrib-group>
      <abstract>
        <p>We present the Swiss Parliaments Corpus (SPC), an automatically aligned Swiss German speech to Standard German text corpus. This first version of the corpus is based on publicly available data of the Bernese cantonal parliament and consists of 293 hours of data. It was created using a novel forced sentence alignment procedure and an alignment quality estimator, which can be used to trade off corpus size and quality. We trained Automatic Speech Recognition (ASR) models as baselines on different subsets of the data and achieved a Word Error Rate (WER) of 0.278 and a BLEU score of 0.586 on the SPC test set. The corpus is freely available for download1.</p>
      </abstract>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1 Introduction</title>
      <p>Swiss German is a family of dialects spoken by
around five million people in Switzerland. It is
different from Standard German regarding
phonetics, vocabulary, morphology, and syntax. Swiss
German is mostly a spoken language. While it is
also used in writing, particularly in informal text
messages, it lacks a standardized writing system.
This leads to difficulties for automated text
processing such as spelling ambiguities and a huge
vocabulary size. For Swiss German ASR, we
therefore focus on end-to-end approaches from Swiss
German speech to Standard German text. This can
be viewed as a Speech Translation problem with
similar source and target languages. For example,
the Swiss German sentence ”Ide Abfahrt hetter de
sa¨chsti Platz beleit” can be translated to the
Standard German sentence ”In der Abfahrt belegte er
den sechsten Platz”. Here, the past tense changes.</p>
      <p>
        Currently, training an ASR model for Swiss
German is challenging due to the lack of public training
data. Only a few hours of Swiss German speech
with Standard German text are available. To reach
high-quality ASR results, a corpus with thousands
of hours of transcribed speech is required. For
example, Park et al. (2020) set the current
stateof-the-art on the English LibriSpeech
        <xref ref-type="bibr" rid="ref9">(Panayotov
et al., 2015)</xref>
        test-other benchmark with a WER of
0.034 using 960 hours of labeled training data and
another 57700 hours of unlabeled data.
      </p>
      <p>While there is no ready-to-use training data,
many Swiss parliaments record their debates. Most
communal and some cantonal parliaments hold
their meetings in Swiss German. Some of them
do a full transcript of the recordings in Standard
German resulting in more than 4000 hours of raw
data.</p>
      <p>To transform the raw data into training data, we
developed a novel forced sentence alignment
algorithm which handles the problems created by the
language mismatch between audio and text such
as changes to the word order within a sentence
(sentence reordering). It is based on a German
ASR model and global alignment and includes a
learned filter component specifically tuned for the
Swiss German speech to Standard German text use
case2. Using the developed alignment algorithm,
we created and published a first corpus called the
Swiss Parliaments Corpus consisting of data from
the parliament Grosser Rat Kanton Bern.</p>
      <p>The remainder of this paper is structured as
follows: Related work is discussed in section 2. The
2The code is available in our GitHub
repository: https://github.com/festivalhopper/
swiss-parliaments-corpus-paper
forced sentence alignment procedure is described
in section 3. Details about our corpus can be found
in section 4. Section 5 contains baseline models
and experiments. Section 6 wraps up the paper and
gives directions for future work.
2</p>
    </sec>
    <sec id="sec-2">
      <title>Related Work</title>
      <p>
        An earlier version of this corpus was previously
published as part of the Low-Resource
Speech-toText shared task at GermEval 2020 (GermEval Task
4)
        <xref ref-type="bibr" rid="ref13">(Plu¨ss et al., 2020)</xref>
        . To our knowledge, there are
only two other publicly available corpora for Swiss
German ASR. ArchiMob (Samardzˇic´ et al., 2016)
includes 69 hours of Swiss German speech and
corresponding Swiss German transcripts.
Unfortunately, Standard German transcripts are not
available. The Radio Rottu Oberwallis dataset (Garner
et al., 2014) includes 8 hours of speech, of which
only 2 hours have Standard German transcripts
in addition to Swiss German transcripts.
Furthermore, the Standard German dataset of the Common
Voice (Ardila et al., 2019) ASR corpus has 1 % of
its utterances spoken in a Swiss German accent,
which however strongly differs from actual Swiss
German speech.
      </p>
      <p>There are different approaches to forced
alignment of long speech recordings in the context of
creating an ASR corpus. Our procedure is similar
to that described in Hazen (2006); Panayotov et al.
(2015); Pratap et al. (2020). Like our approach,
these methods initially transcribe audios using an
ASR system, followed by an alignment stage and
a final refinement stage. The main differences are
that these approaches do not yield strictly
sentencelevel alignments, which are a requirement for our
work due to the possibility of sentence reorderings
between Standard German and Swiss German, and
our novel approach to filter the corpus and improve
the quality.
3</p>
    </sec>
    <sec id="sec-3">
      <title>Forced Sentence Alignment Procedure</title>
      <p>
        Our forced sentence alignment procedure takes a
Swiss German recording of arbitrary length and the
corresponding manual Standard German transcript
as inputs. The audio file is transcribed with an
ASR model. An important requirement for this
model is the ability to annotate accurate start and
end times of each word in the output. Since no
publicly available Swiss German ASR model with
this feature exists, we resort to a Standard German
model. The ASR transcript is then globally aligned
to the manual transcript using the Biopython (Cock
et al., 2009) implementation of the
NeedlemanWunsch algorithm
        <xref ref-type="bibr" rid="ref8">(Needleman and Wunsch, 1970)</xref>
        .
The manual transcript is split into sentences using
spaCy
        <xref ref-type="bibr" rid="ref16 ref3">(Honnibal and Montani, 2017)</xref>
        . Each of
these sentences is mapped to a start and end time
in the recording via the global alignment and the
per-word start and end times provided by the ASR
model.
3.1
      </p>
      <sec id="sec-3-1">
        <title>Alignment Corpus and Metrics</title>
        <p>We created a separate internal alignment corpus
to be able to measure the quality of our sentence
alignment. It consists of almost 6 hours of
transcribed recordings from four different parliaments
and one other data source. We split the corpus into
a training and a test set (60-40 split). Recordings
and transcripts were manually sentence-aligned.</p>
        <p>We define an aligned sentence as a three-tuple
of the sentence, its start and end time. We call an
aligned sentence empty if the start and end times
are not set, which means the sentence is not spoken
in the recording. This happens because transcripts
sometimes have errors such as missing or
additional sentences.</p>
        <p>Our goal is to maximize the following metrics
during the creation of the corpus:
• The Intersection over Union (IoU) reflects
the alignment quality. We report the mean IoU
over all predicted aligned sentences for which
the manual as well as the predicted aligned
sentence are not empty.
• The sentence precision and recall reflect the
corpus quality and size. A predicted aligned
sentence counts as true positive (TP) if the
manual as well as the predicted aligned
sentence are not empty. True negative (TN)
means the manual as well as the predicted
aligned sentence are empty. False positive
(FP) means the manual aligned sentence is
empty, but the predicted aligned sentence is
not empty. False negative (FN) means the
manual aligned sentence is not empty, but the
predicted aligned sentence is empty. The
sentence precision is equal to TP / (TP + FP). The
sentence recall is equal to TP / (TP + FN).</p>
      </sec>
      <sec id="sec-3-2">
        <title>3.2 IoU Estimate Filter and Further</title>
      </sec>
      <sec id="sec-3-3">
        <title>Refinements</title>
        <p>We filter out sentences with a bad alignment quality
based on an estimate of their IoU. We fit a Gradient</p>
      </sec>
      <sec id="sec-3-4">
        <title>Hyperparameter</title>
        <p>num leaves
min child samples
max bin</p>
      </sec>
      <sec id="sec-3-5">
        <title>Value</title>
        <p>Boosting regressor to estimate a sentence’s IoU
using the following features:
• Length ratio of the manual transcript
sentence to the part of the ASR transcript it was
aligned to
• Alignment score of the manual transcript
sentence, normalized by its length
• Mean speech recognition confidence as
reported by the ASR system over the words the
manual transcript sentence was aligned to
• Chars per second, i.e. ratio of the manual
transcript sentence length to the audio length
(predicted aligned sentence end time minus
start time)</p>
        <p>We use the LightGBM implementation by Ke
et al. (2017). Table 1 shows our hyperparameters.
These were found using Bayesian optimization.</p>
        <p>The regressor estimates the IoU in a 3-fold cross
validation experiment on the training set of our
alignment corpus with a mean absolute error of
0.108 (IoU values are in the interval [0, 1]). We
propose two different IoU estimate thresholds. A
threshold of 0.7 is supposed to keep as many
sentences as possible and only discard sentences with
a bad alignment quality, e.g. for a training set.
A threshold of 0.9 is supposed to keep only
sentences with a very good alignment quality, e.g. for
a test set. Thresholds were found using a parameter
sweep on the test set of the alignment corpus.</p>
        <p>Two more refinements were implemented: to
filter out manual transcripts that are clearly
mismatched or incomplete, no alignment is created
if the length ratio of the longer transcript to the
shorter transcript is greater than six. The optimal
ratio was found using a parameter sweep on the
alignment corpus test set. Finally, we fit a start and
and end time correction offset on the training set
of our alignment corpus and calibrate the start and</p>
        <sec id="sec-3-5-1">
          <title>Amazon Transcribe 0:626 Google Speech-to-Text 0:725</title>
          <p>end times of each sentence by adding the
correction offset. This leads to a minor IoU improvement
because the times reported by the ASR model can
be slightly off.
3.3</p>
        </sec>
      </sec>
      <sec id="sec-3-6">
        <title>Experiments and Results</title>
        <p>
          We evaluated two ASR models, Amazon
Transcribe34 and Google Speech-to-Text56, on the
public test set of GermEval 2020 Task 4
          <xref ref-type="bibr" rid="ref13">(Plu¨ss et al.,
2020)</xref>
          . Table 2 shows the results of this
comparison. Amazon is ahead of Google by 0.1 WER.
This suggests that Amazon succeeded in
improving the performance for Swiss German with its
specialized model, but still has a long way to go
to achieve a general-purpose ASR model with a
WER comparable to English or Standard German
models. In comparison to the performance of the
winning contribution to GermEval 2020 Task 4 by
Bu¨chi et al. (2020) with a WER of 0.403, Amazon
Transcribe is more than 0.2 WER behind.
        </p>
        <p>We conducted experiments using Amazon
Transcribe and Google Speech-to-Text as the ASR
engines with different combinations of refinements7.
We determined the parameters for the global
alignment algorithm using Bayesian optimization with
3-fold cross validation on the training set of our
alignment corpus. They are listed in appendix A.
Table 3 shows the results. The lead in WER by 0.1
(see table 2) for Amazon translates to a mean IoU
that is 0.151 higher than Google’s result with the
same settings. The 0.045 advantage of the latter
in sentence recall does not make up for this, even
less so because we prefer quality over quantity. For
3https://aws.amazon.com/transcribe
4We used the ”Swiss German” model. Based on the
information we found, we believe this is a model for Standard
German, specialized on Swiss accents, not for actual Swiss
German.</p>
        <p>5https://cloud.google.com/
speech-to-text</p>
        <p>6We used the ”German (Germany)” model, ”German
(Switzerland)” was not yet available at the time of the
experiment.</p>
        <p>7We could not use the Bu¨chi et al. model because it does
not provide word start and end times</p>
      </sec>
      <sec id="sec-3-7">
        <title>Audio Length in Hours</title>
      </sec>
      <sec id="sec-3-8">
        <title>Number of Speakers</title>
        <sec id="sec-3-8-1">
          <title>Amazon Transcribe</title>
          <p>Amazon Transcribe
Amazon Transcribe
Amazon Transcribe
Google Speech-to-Text
Amazon Transcribe</p>
        </sec>
        <sec id="sec-3-8-2">
          <title>Amazon Transcribe</title>
          <p>No Refinements
Length Ratio
Time Calibration
Length Ratio + Time Calibration
Length Ratio + Time Calibration
Length Ratio + Time Calibration
+ IoU Estimate Filter 0.7
Length Ratio + Time Calibration
+ IoU Estimate Filter 0.9
Amazon Transcribe, enabling length ratio filtering
as well as time calibration appears to be the best
option, resulting in a mean IoU of 0.840 and a
sentence recall of 0.949. The alignment quality can
be further improved using the IoU estimate filter.
A threshold of 0.7 leads to an increase of 0.048 in
mean IoU and a decrease of 0.127 in sentence
recall, whereas a threshold of 0.9 leads to an increase
of 0.087 in mean IoU and a decrease of 0.461 in
sentence recall. The sentence precision is perfect
in all experiments.
4</p>
        </sec>
      </sec>
    </sec>
    <sec id="sec-4">
      <title>Swiss Parliaments Corpus</title>
      <p>Using our forced sentence alignment procedure, we
created and published8 a corpus called the Swiss
Parliaments Corpus. It is based on recordings and
transcripts from the parliament Grosser Rat Kanton
Bern9. As expected, given the location of the
parliament, most speakers have a Bernese dialect. The
recordings are MP4 videos, one video per
parliament meeting, with a length spanning from 28
minutes to 4 hours and 2 minutes. The transcripts are
in PDF format, with one PDF containing a whole
session with usually around 10 to 15 meetings.
4.1</p>
      <sec id="sec-4-1">
        <title>Corpus Parts</title>
        <p>Table 4 gives an overview of the different corpus
parts and their sizes. We created an unfiltered
training set called train all with 293 hours of data. We
then used IoU estimate filtering to create two
training subsets, train 0.7 with a threshold of 0.7 and
8https://www.cs.technik.fhnw.ch/
i4ds-datasets</p>
        <p>9https://www.gr.be.ch/gr/de/index/
sessionen/sessionen.html</p>
      </sec>
      <sec id="sec-4-2">
        <title>Corpus Part</title>
        <p>Raw data
train all
train 0.7
train 0.9
test
256 hours of data as well as train 0.9 with a
threshold of 0.9 and 176 hours of data. The unfiltered
training set contains an IoU estimate column to
create a training set with a custom threshold. The test
set was created with a threshold of 0.9 and contains
6 hours of data. We could therefore transform 65
% of the raw data to training or test data.</p>
      </sec>
      <sec id="sec-4-3">
        <title>4.2 Settings, Filters, Split</title>
        <p>We used semi-global alignment parameters (no gap
penalties on the start and end of both sequences,
see appendix B) to deal with incomplete recordings
and additional irrelevant recorded audio. Length
ratio filtering was disabled while time calibration
was enabled. We applied the following additional
filters:
• Chars per second must be between 6 and 23.</p>
        <p>The average for chars per second in this
corpus is 15. Aligned sentences outside of this
range probably either contain a lot of idle time
in the recording or additional text that is not
recorded.
• We detect the language of each sentence using
langdetect10 and only keep German sentences.
• (Test set only) Audio length must be at least 1
second.
• (Test set only) Audio length must be less than
15 seconds.
• (Test set only) Sentences must be unique
across the whole dataset.</p>
        <p>Speakers are automatically deduplicated. The
traintest split guarantees that the utterances of a speaker
are only contained in either the training set or the
test set, never in both. To ensure that the speakers
in the test set are diverse enough, a speaker can
only be part of the test set if her or his utterances
make up less than 10 % of the whole test set.
5</p>
      </sec>
    </sec>
    <sec id="sec-5">
      <title>ASR Baselines</title>
      <p>
        All the baseline models are implemented using the
ESPnet framework
        <xref ref-type="bibr" rid="ref17">(Watanabe et al., 2018)</xref>
        . We
trained a Transformer model (Vaswani et al., 2017)
as well as a Conformer model
        <xref ref-type="bibr" rid="ref1">(Gulati et al., 2020)</xref>
        .
The network architectures closely follow the
Common Voice example of ESPnet11 using a hybrid
CTC/attention encoder-decoder framework
        <xref ref-type="bibr" rid="ref18">(Watanabe et al., 2017)</xref>
        .
      </p>
      <p>Inputs are first down-sampled to 1/4 length by
two strided 2D convolution layers and ReLU
activations. The Transformer encoder consists of 12
self-attention blocks with 2048 units. Similarly,
the Conformer encoder uses 12 Conformer-layers
with 2048 units. Both Transformer and Conformer
models use a Transformer decoder with six
selfattention blocks with 2048 units.</p>
      <p>
        For the training of both models we use the Adam
optimizer
        <xref ref-type="bibr" rid="ref5 ref6 ref9">(Kingma and Ba, 2015)</xref>
        and a warmup
learning rate schedule similar to the one proposed
in Vaswani et al. (2017), but with a fixed warmup
period of 25000 steps and a maximum learning rate
of 0.002. As input we use 80-channel log-mel
filterbanks that are shifted to have a mean of zero. Speed
perturbation
        <xref ref-type="bibr" rid="ref6">(Ko et al., 2015)</xref>
        with random factors
between 0.9 and 1.1 and SpecAugment
        <xref ref-type="bibr" rid="ref11">(Park et al.,
2019)</xref>
        are used for data augmentation. Both models
are trained with a combined Connectionist
Temporal Classification (CTC) (Graves et al., 2006)
10https://pypi.org/project/langdetect
11https://github.com/espnet/espnet/
tree/master/egs2/commonvoice/asr1
      </p>
      <sec id="sec-5-1">
        <title>Model</title>
      </sec>
      <sec id="sec-5-2">
        <title>Dataset</title>
        <sec id="sec-5-2-1">
          <title>Transformer train all train 0.7 train 0.9</title>
        </sec>
        <sec id="sec-5-2-2">
          <title>Conformer train all train 0.7 train 0.9</title>
          <p>0:297
0:293
0:303
0:289
0:278
0:287
0:548
0:553
0:537
0:577
0:586
0:577
and cross entropy loss with weights 0.3 and 0.7,
respectively.</p>
          <p>
            During decoding we use a 16-layer Transformer
language model, trained on the SPC texts as well as
the German texts of the EuroParl Corpus
            <xref ref-type="bibr" rid="ref7">(Koehn,
2005)</xref>
            , using beam search with a beam size of 50.
          </p>
          <p>
            We trained both models on all subsets of the SPC
until convergence for 200 epochs. The results of
all models are shown in Table 5. We report WER,
commonly used to evaluate ASR systems, as well
as the BLEU
            <xref ref-type="bibr" rid="ref10">(Papineni et al., 2002)</xref>
            score,
commonly used to evaluate Machine Translation and
Speech Translation systems. For BLEU, we use
the implementation provided by NLTK (Bird et al.,
2009) with default parameters. In our experiments,
WER and BLEU show a negative correlation as
expected (WER: lower is better, BLEU: higher is
better), indicating that both are similarly useful
metrics.
          </p>
          <p>The Conformer model performs better than the
Transformer model in all experiments. This is in
line with the findings of Gulati et al. (2020) on
the LibriSpeech benchmark. For both models, all
dataset splits resulted in similar WER and BLEU
scores, even though their sizes differ significantly
(see Table 4). This can be explained by the higher
quality when filtering based on IoU scores. As a
result, we can train models on 60 % of the data,
resulting in faster training, without losing
performance.
6</p>
        </sec>
      </sec>
    </sec>
    <sec id="sec-6">
      <title>Conclusion</title>
      <p>In this work, we introduced the Swiss Parliament
Corpus, an automatically aligned Swiss German
speech to Standard German text corpus. We
proposed a multi-stage forced sentence alignment
procedure that leverages existing Standard German
ASR systems and uses a novel IoU estimator for
refinement. We also provided Transformer and
Conformer ASR baseline models that showcase
the benefits of the IoU estimates. The best model
achieves a WER of 0.278 and a BLEU score of
0.586 on the SPC test set.</p>
      <p>We believe that our forced sentence
alignment procedure is a step towards making
largevocabulary speech recognition for all Swiss
German dialects possible. The SPC with its 293 hours
of training data supports this thesis. It is freely
available for download12.</p>
      <p>In future work, we plan to increase the corpus
size and the dialect diversity by aligning recordings
and transcripts of additional parliaments. We also
plan to collect data for a test set representing all
Swiss German dialects since the SPC is
domainspecific and includes mostly Bernese speakers.
This would facilitate a fair comparison for Swiss
German ASR systems. In the future, we will also
use the trained models to improve the forced
sentence alignment algorithm results in a similar way
as Sennrich and Volk (2011).</p>
      <p>Furthermore, the effects of the IoU filter on the
quality of the dataset as well as the impact on the
model need further investigation. Finally, we want
to investigate the correlation of WER and BLEU
with human evaluation to understand which metric
is most appropriate for the problem. In this context,
it would also be interesting to further investigate
the differences between a literal transcription in
Swiss German and a Standard German translation.</p>
    </sec>
    <sec id="sec-7">
      <title>Acknowledgments</title>
      <p>First and foremost, we would like to thank the
parliamentary services of the canton of Bern for
their work on the transcription of the debates and
for publishing recordings and transcripts on their
website. Without them, the SPC would not exist.</p>
      <p>Furthermore, we thank Pascal Thormeier who
created the first version of our alignment corpus
during his bachelor’s thesis.</p>
      <p>We also thank the participants of GermEval 2020
Task 4 for the fruitful discussions in the aftermath
of the task, which lead to several improvements of
the SPC.
Rosana Ardila, Megan Branson, Kelly Davis, Michael
Henretty, Michael Kohler, Josh Meyer, Reuben
12https://www.cs.technik.fhnw.ch/
i4ds-datasets
Morais, Lindsay Saunders, Francis M Tyers, and
Gregor Weber. 2019. Common voice: A
massivelymultilingual speech corpus. arXiv preprint
arXiv:1912.06670.</p>
      <p>Steven Bird, Ewan Klein, and Edward Loper. 2009.</p>
      <p>Natural language processing with Python:
analyzing text with the natural language toolkit. ” O’Reilly
Media, Inc.”.</p>
      <p>Matthias Bu¨chi, Malgorzata Anna Ulasik, Manuela
Hu¨rlimann, Fernando Benites, Pius von Da¨niken,
and Mark Cieliebak. 2020. Zhaw-init at germeval
2020 task 4: Low-resource speech-to-text. In
SWISSTEXT &amp; KONVENS 2020, Proceedings of the 5th
Swiss Text Analytics Conference (SwissText) &amp;
16th Conference on Natural Language Processing
(KONVENS).</p>
      <p>Peter J. A. Cock, Tiago Antao, Jeffrey T. Chang,
Brad A. Chapman, Cymon J. Cox, Andrew Dalke,
Iddo Friedberg, Thomas Hamelryck, Frank Kauff,
Bartek Wilczynski, and Michiel J. L. de Hoon.
2009. Biopython: freely available Python tools for
computational molecular biology and
bioinformatics. Bioinformatics, 25(11):1422–1423.</p>
      <p>Philip Garner, David Imseng, and Thomas Meyer. 2014.</p>
      <p>Automatic speech recognition and translation of a
swiss german dialect: Walliserdeutsch. In
Proceedings of the Annual Conference of the
International Speech Communication Association,
INTERSPEECH.</p>
      <p>Alex Graves, Santiago Ferna´ndez, Faustino Gomez,
and Ju¨rgen Schmidhuber. 2006. Connectionist
temporal classification: labelling unsegmented
sequence data with recurrent neural networks. In
Proceedings of the 23rd international conference on
Machine learning, pages 369–376.
182, Riga, Latvia. Northern European Association
for Language Technology (NEALT).</p>
      <p>A</p>
    </sec>
    <sec id="sec-8">
      <title>Alignment Parameters Optimized on</title>
    </sec>
    <sec id="sec-9">
      <title>Alignment Corpus</title>
      <p>Alignment Parameter
match score
mismatch score
truth left open gap score
truth internal open gap score
truth right open gap score
truth left extend gap score
truth internal extend gap score
truth right extend gap score
stt left open gap score
stt internal open gap score
stt right open gap score
stt left extend gap score
stt internal extend gap score
stt right extend gap score
Value
0:039
1:000
0:504
1:000
0:440
0:244
0:482
0:259
1:000
0:770
0:982
0:253
0:770
0:562</p>
    </sec>
    <sec id="sec-10">
      <title>Alignment Parameters for SPC</title>
    </sec>
  </body>
  <back>
    <ref-list>
      <ref id="ref1">
        <mixed-citation>
          <string-name>
            <given-names>Anmol</given-names>
            <surname>Gulati</surname>
          </string-name>
          , James Qin, Chung-Cheng Chiu, Niki Parmar, Yu Zhang, Jiahui Yu, Wei Han,
          <string-name>
            <surname>Shibo</surname>
            <given-names>Wang</given-names>
          </string-name>
          , Zhengdong Zhang, Yonghui Wu, and
          <string-name>
            <given-names>Ruoming</given-names>
            <surname>Pang</surname>
          </string-name>
          .
          <year>2020</year>
          .
          <article-title>Conformer: Convolution-augmented Transformer for Speech Recognition</article-title>
          .
          <source>In Proceedings of Interspeech</source>
          , pages
          <fpage>5036</fpage>
          -
          <lpage>5040</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref2">
        <mixed-citation>
          <string-name>
            <given-names>Timothy J.</given-names>
            <surname>Hazen</surname>
          </string-name>
          .
          <year>2006</year>
          .
          <article-title>Automatic alignment and error correction of human generated transcripts for long speech recordings</article-title>
          .
          <source>In Proceedings of Interspeech.</source>
        </mixed-citation>
      </ref>
      <ref id="ref3">
        <mixed-citation>
          <string-name>
            <given-names>Matthew</given-names>
            <surname>Honnibal</surname>
          </string-name>
          and
          <string-name>
            <given-names>Ines</given-names>
            <surname>Montani</surname>
          </string-name>
          .
          <year>2017</year>
          .
          <article-title>spaCy 2: Natural language understanding with Bloom embeddings, convolutional neural networks and incremental parsing</article-title>
          . To appear.
        </mixed-citation>
      </ref>
      <ref id="ref4">
        <mixed-citation>
          <string-name>
            <given-names>Guolin</given-names>
            <surname>Ke</surname>
          </string-name>
          , Qi Meng, Thomas Finley, Taifeng Wang,
          <string-name>
            <surname>Wei</surname>
            <given-names>Chen</given-names>
          </string-name>
          , Weidong Ma, Qiwei Ye, and
          <string-name>
            <surname>Tie-Yan Liu</surname>
          </string-name>
          .
          <year>2017</year>
          .
          <article-title>Lightgbm: A highly efficient gradient boosting decision tree</article-title>
          .
          <source>In Proceedings of the 31st International Conference on Neural Information Processing Systems</source>
          , page
          <fpage>3149</fpage>
          -
          <lpage>3157</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref5">
        <mixed-citation>
          <string-name>
            <given-names>Diederik P.</given-names>
            <surname>Kingma</surname>
          </string-name>
          and
          <string-name>
            <given-names>Jimmy</given-names>
            <surname>Ba</surname>
          </string-name>
          .
          <year>2015</year>
          .
          <article-title>Adam: A method for stochastic optimization</article-title>
          .
          <source>In 3rd International Conference on Learning Representations</source>
          ,
          <string-name>
            <surname>ICLR</surname>
          </string-name>
          <year>2015</year>
          .
        </mixed-citation>
      </ref>
      <ref id="ref6">
        <mixed-citation>
          <string-name>
            <given-names>Tom</given-names>
            <surname>Ko</surname>
          </string-name>
          , Vijayaditya Peddinti, Daniel Povey, and
          <string-name>
            <given-names>Sanjeev</given-names>
            <surname>Khudanpur</surname>
          </string-name>
          .
          <year>2015</year>
          .
          <article-title>Audio augmentation for speech recognition</article-title>
          .
          <source>In Sixteenth Annual Conference of the International Speech Communication Association.</source>
        </mixed-citation>
      </ref>
      <ref id="ref7">
        <mixed-citation>
          <string-name>
            <given-names>Philipp</given-names>
            <surname>Koehn</surname>
          </string-name>
          .
          <year>2005</year>
          .
          <article-title>Europarl: A parallel corpus for statistical machine translation</article-title>
          .
          <source>In MT summit</source>
          , volume
          <volume>5</volume>
          , pages
          <fpage>79</fpage>
          -
          <lpage>86</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref8">
        <mixed-citation>
          <string-name>
            <surname>Saul</surname>
            <given-names>B.</given-names>
          </string-name>
          <string-name>
            <surname>Needleman</surname>
            and
            <given-names>Christian D.</given-names>
          </string-name>
          <string-name>
            <surname>Wunsch</surname>
          </string-name>
          .
          <year>1970</year>
          .
          <article-title>A general method applicable to the search for similarities in the amino acid sequence of two proteins</article-title>
          .
          <source>J. Mol. Biol</source>
          .,
          <volume>48</volume>
          :
          <fpage>443</fpage>
          -
          <lpage>453</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref9">
        <mixed-citation>
          <string-name>
            <given-names>Vassil</given-names>
            <surname>Panayotov</surname>
          </string-name>
          , Guoguo Chen, Daniel Povey, and
          <string-name>
            <given-names>Sanjeev</given-names>
            <surname>Khudanpur</surname>
          </string-name>
          .
          <year>2015</year>
          .
          <article-title>Librispeech: An ASR corpus based on public domain audio books</article-title>
          .
          <source>In 2015 IEEE International Conference on Acoustics, Speech and Signal Processing</source>
          , ICASSP, pages
          <fpage>5206</fpage>
          -
          <lpage>5210</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref10">
        <mixed-citation>
          <string-name>
            <given-names>Kishore</given-names>
            <surname>Papineni</surname>
          </string-name>
          , Salim Roukos, Todd Ward, and
          <string-name>
            <given-names>WeiJing</given-names>
            <surname>Zhu</surname>
          </string-name>
          .
          <year>2002</year>
          .
          <article-title>Bleu: a method for automatic evaluation of machine translation</article-title>
          .
          <source>In Proceedings of the 40th Annual Meeting of the Association for Computational Linguistics</source>
          , pages
          <fpage>311</fpage>
          -
          <lpage>318</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref11">
        <mixed-citation>
          <string-name>
            <surname>Daniel S Park</surname>
            ,
            <given-names>William</given-names>
          </string-name>
          <string-name>
            <surname>Chan</surname>
          </string-name>
          , Yu Zhang, Chung-Cheng Chiu, Barret Zoph,
          <string-name>
            <surname>Ekin D Cubuk</surname>
          </string-name>
          , and
          <string-name>
            <surname>Quoc</surname>
            <given-names>V</given-names>
          </string-name>
          <string-name>
            <surname>Le</surname>
          </string-name>
          .
          <year>2019</year>
          .
          <article-title>Specaugment: A simple data augmentation method for automatic speech recognition</article-title>
          . arXiv preprint arXiv:
          <year>1904</year>
          .08779.
        </mixed-citation>
      </ref>
      <ref id="ref12">
        <mixed-citation>
          <string-name>
            <given-names>Daniel S.</given-names>
            <surname>Park</surname>
          </string-name>
          , Yu Zhang, Ye Jia, Wei Han, ChungCheng Chiu,
          <string-name>
            <given-names>Bo</given-names>
            <surname>Li</surname>
          </string-name>
          ,
          <string-name>
            <surname>Yonghui Wu</surname>
          </string-name>
          , and
          <string-name>
            <surname>Quoc</surname>
            <given-names>V</given-names>
          </string-name>
          <string-name>
            <surname>Le</surname>
          </string-name>
          .
          <year>2020</year>
          .
          <article-title>Improved noisy student training for automatic speech recognition</article-title>
          .
          <source>In Proceedings of Interspeech.</source>
        </mixed-citation>
      </ref>
      <ref id="ref13">
        <mixed-citation>
          <string-name>
            <given-names>Michel</given-names>
            <surname>Plu</surname>
          </string-name>
          <article-title>¨ss, Lukas Neukom</article-title>
          , and
          <string-name>
            <given-names>Manfred</given-names>
            <surname>Vogel</surname>
          </string-name>
          .
          <year>2020</year>
          .
          <article-title>Germeval 2020 task 4: Low-resource speechto-text</article-title>
          .
          <source>In SWISSTEXT &amp; KONVENS 2020, Proceedings of the 5th Swiss Text Analytics Conference (SwissText) &amp; 16th Conference on Natural Language Processing (KONVENS).</source>
        </mixed-citation>
      </ref>
      <ref id="ref14">
        <mixed-citation>
          <string-name>
            <given-names>Vineel</given-names>
            <surname>Pratap</surname>
          </string-name>
          , Qiantong Xu, Anuroop Sriram, Gabriel Synnaeve, and
          <string-name>
            <given-names>Ronan</given-names>
            <surname>Collobert</surname>
          </string-name>
          .
          <year>2020</year>
          .
          <article-title>Mls: A largescale multilingual dataset for speech research</article-title>
          . arXiv preprint arXiv:
          <year>2012</year>
          .03411.
        </mixed-citation>
      </ref>
      <ref id="ref15">
        <mixed-citation>
          <string-name>
            <surname>Tanja</surname>
            <given-names>Samardzˇic´</given-names>
          </string-name>
          , Yves Scherrer, and
          <string-name>
            <given-names>Elvira</given-names>
            <surname>Glaser</surname>
          </string-name>
          .
          <year>2016</year>
          .
          <article-title>Archimob - a corpus of spoken swiss german</article-title>
          .
          <source>Proceedings of the Tenth International Conference on Language Resources and Evaluation (LREC</source>
          <year>2016</year>
          ), pages
          <fpage>4061</fpage>
          -
          <lpage>4066</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref16">
        <mixed-citation>
          <string-name>
            <given-names>Rico</given-names>
            <surname>Sennrich</surname>
          </string-name>
          and
          <string-name>
            <given-names>Martin</given-names>
            <surname>Volk</surname>
          </string-name>
          .
          <year>2011</year>
          .
          <article-title>Iterative, MTbased sentence alignment of parallel texts</article-title>
          .
          <source>In Proceedings of the 18th Nordic Conference of Computational Linguistics (NODALIDA</source>
          <year>2011</year>
          ), pages
          <fpage>175</fpage>
          -
          <string-name>
            <surname>Ashish</surname>
            <given-names>Vaswani</given-names>
          </string-name>
          , Noam Shazeer, Niki Parmar, Jakob Uszkoreit, Llion Jones, Aidan N Gomez,
          <article-title>Ł ukasz Kaiser, and</article-title>
          <string-name>
            <given-names>Illia</given-names>
            <surname>Polosukhin</surname>
          </string-name>
          .
          <year>2017</year>
          .
          <article-title>Attention is all you need</article-title>
          .
          <source>In Advances in Neural Information Processing Systems</source>
          , volume
          <volume>30</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref17">
        <mixed-citation>
          <string-name>
            <given-names>Shinji</given-names>
            <surname>Watanabe</surname>
          </string-name>
          , Takaaki Hori, Shigeki Karita, Tomoki Hayashi, Jiro Nishitoba, Yuya Unno, Nelson Enrique Yalta Soplin, Jahn Heymann, Matthew Wiesner, Nanxin Chen, Adithya Renduchintala, and
          <string-name>
            <given-names>Tsubasa</given-names>
            <surname>Ochiai</surname>
          </string-name>
          .
          <year>2018</year>
          .
          <article-title>ESPnet: End-to-end speech processing toolkit</article-title>
          .
          <source>In Proceedings of Interspeech</source>
          , pages
          <fpage>2207</fpage>
          -
          <lpage>2211</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref18">
        <mixed-citation>
          <string-name>
            <given-names>Shinji</given-names>
            <surname>Watanabe</surname>
          </string-name>
          , Takaaki Hori, Suyoun Kim, John R Hershey, and
          <string-name>
            <given-names>Tomoki</given-names>
            <surname>Hayashi</surname>
          </string-name>
          .
          <year>2017</year>
          .
          <article-title>Hybrid ctc/attention architecture for end-to-end speech recognition</article-title>
          .
          <source>IEEE Journal of Selected Topics in Signal Processing</source>
          ,
          <volume>11</volume>
          (
          <issue>8</issue>
          ):
          <fpage>1240</fpage>
          -
          <lpage>1253</lpage>
          .
        </mixed-citation>
      </ref>
    </ref-list>
  </back>
</article>