473 research outputs found
Unsupervised crosslingual adaptation of tokenisers for spoken language recognition
Phone tokenisers are used in spoken language recognition (SLR) to obtain elementary
phonetic information. We present a study on the use of deep neural
network tokenisers. Unsupervised crosslingual adaptation was performed to
adapt the baseline tokeniser trained on English conversational telephone speech
data to different languages. Two training and adaptation approaches, namely
cross-entropy adaptation and state-level minimum Bayes risk adaptation, were
tested in a bottleneck i-vector and a phonotactic SLR system. The SLR systems
using the tokenisers adapted to different languages were combined using score
fusion, giving 7-18% reduction in minimum detection cost function (minDCF)
compared with the baseline configurations without adapted tokenisers. Analysis
of results showed that the ensemble tokenisers gave diverse representation of
phonemes, thus bringing complementary effects when SLR systems with different
tokenisers were combined. SLR performance was also shown to be related
to the quality of the adapted tokenisers
Unsupervised crosslingual adaptation of tokenisers for spoken language recognition
Phone tokenisers are used in spoken language recognition (SLR) to obtain elementary
phonetic information. We present a study on the use of deep neural
network tokenisers. Unsupervised crosslingual adaptation was performed to
adapt the baseline tokeniser trained on English conversational telephone speech
data to different languages. Two training and adaptation approaches, namely
cross-entropy adaptation and state-level minimum Bayes risk adaptation, were
tested in a bottleneck i-vector and a phonotactic SLR system. The SLR systems
using the tokenisers adapted to different languages were combined using score
fusion, giving 7-18% reduction in minimum detection cost function (minDCF)
compared with the baseline configurations without adapted tokenisers. Analysis
of results showed that the ensemble tokenisers gave diverse representation of
phonemes, thus bringing complementary effects when SLR systems with different
tokenisers were combined. SLR performance was also shown to be related
to the quality of the adapted tokenisers
Sequence Teacher-Student Training of Acoustic Models for Automatic Free Speaking Language Assessment
A high performance automatic speech recognition (ASR) system is
an important constituent component of an automatic language assessment system for free speaking language tests. The ASR system
is required to be capable of recognising non-native spontaneous English
speech and to be deployable under real-time conditions. The
performance of ASR systems can often be significantly improved by
leveraging upon multiple systems that are complementary, such as an
ensemble. Ensemble methods, however, can be computationally expensive,
often requiring multiple decoding runs, which makes them
impractical for deployment. In this paper, a lattice-free implementation
of sequence-level teacher-student training is used to reduce this
computational cost, thereby allowing for real-time applications. This
method allows a single student model to emulate the performance of
an ensemble of teachers, but without the need for multiple decoding
runs. Adaptations of the student model to speakers from different
first languages (L1s) and grades are also explored.Cambridge Assessment Englis
Automatic speaker recognition
06.03.2018 tarihli ve 30352 sayılı Resmi Gazetede yayımlanan “Yükseköğretim Kanunu İle Bazı Kanun Ve Kanun Hükmünde Kararnamelerde Değişiklik Yapılması Hakkında Kanun” ile 18.06.2018 tarihli “Lisansüstü Tezlerin Elektronik Ortamda Toplanması, Düzenlenmesi ve Erişime Açılmasına İlişkin Yönerge” gereğince tam metin erişime açılmıştır
Joint Morphological and Syntactic Disambiguation
In morphologically rich languages, should morphological and syntactic disambiguation be treated sequentially or as a single problem? We describe several efficient, probabilistically interpretable ways to apply joint inference to morphological and syntactic disambiguation using lattice parsing. Joint inference is shown to compare favorably to pipeline parsing methods across a variety of component models. State-of-the-art performance on Hebrew Treebank parsing is demonstrated using the new method. The benefits of joint inference are modest with the current component models, but appear to increase as components themselves improve
A Speaker Verification Backend with Robust Performance across Conditions
In this paper, we address the problem of speaker verification in conditions
unseen or unknown during development. A standard method for speaker
verification consists of extracting speaker embeddings with a deep neural
network and processing them through a backend composed of probabilistic linear
discriminant analysis (PLDA) and global logistic regression score calibration.
This method is known to result in systems that work poorly on conditions
different from those used to train the calibration model. We propose to modify
the standard backend, introducing an adaptive calibrator that uses duration and
other automatically extracted side-information to adapt to the conditions of
the inputs. The backend is trained discriminatively to optimize binary
cross-entropy. When trained on a number of diverse datasets that are labeled
only with respect to speaker, the proposed backend consistently and, in some
cases, dramatically improves calibration, compared to the standard PLDA
approach, on a number of held-out datasets, some of which are markedly
different from the training data. Discrimination performance is also
consistently improved. We show that joint training of the PLDA and the adaptive
calibrator is essential -- the same benefits cannot be achieved when freezing
PLDA and fine-tuning the calibrator. To our knowledge, the results in this
paper are the first evidence in the literature that it is possible to develop a
speaker verification system with robust out-of-the-box performance on a large
variety of conditions
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