4,162 research outputs found
Multitask Learning with Low-Level Auxiliary Tasks for Encoder-Decoder Based Speech Recognition
End-to-end training of deep learning-based models allows for implicit
learning of intermediate representations based on the final task loss. However,
the end-to-end approach ignores the useful domain knowledge encoded in explicit
intermediate-level supervision. We hypothesize that using intermediate
representations as auxiliary supervision at lower levels of deep networks may
be a good way of combining the advantages of end-to-end training and more
traditional pipeline approaches. We present experiments on conversational
speech recognition where we use lower-level tasks, such as phoneme recognition,
in a multitask training approach with an encoder-decoder model for direct
character transcription. We compare multiple types of lower-level tasks and
analyze the effects of the auxiliary tasks. Our results on the Switchboard
corpus show that this approach improves recognition accuracy over a standard
encoder-decoder model on the Eval2000 test set
Personalized Acoustic Modeling by Weakly Supervised Multi-Task Deep Learning using Acoustic Tokens Discovered from Unlabeled Data
It is well known that recognizers personalized to each user are much more
effective than user-independent recognizers. With the popularity of smartphones
today, although it is not difficult to collect a large set of audio data for
each user, it is difficult to transcribe it. However, it is now possible to
automatically discover acoustic tokens from unlabeled personal data in an
unsupervised way. We therefore propose a multi-task deep learning framework
called a phoneme-token deep neural network (PTDNN), jointly trained from
unsupervised acoustic tokens discovered from unlabeled data and very limited
transcribed data for personalized acoustic modeling. We term this scenario
"weakly supervised". The underlying intuition is that the high degree of
similarity between the HMM states of acoustic token models and phoneme models
may help them learn from each other in this multi-task learning framework.
Initial experiments performed over a personalized audio data set recorded from
Facebook posts demonstrated that very good improvements can be achieved in both
frame accuracy and word accuracy over popularly-considered baselines such as
fDLR, speaker code and lightly supervised adaptation. This approach complements
existing speaker adaptation approaches and can be used jointly with such
techniques to yield improved results.Comment: 5 pages, 5 figures, published in IEEE ICASSP 201
Quaternion Convolutional Neural Networks for End-to-End Automatic Speech Recognition
Recently, the connectionist temporal classification (CTC) model coupled with
recurrent (RNN) or convolutional neural networks (CNN), made it easier to train
speech recognition systems in an end-to-end fashion. However in real-valued
models, time frame components such as mel-filter-bank energies and the cepstral
coefficients obtained from them, together with their first and second order
derivatives, are processed as individual elements, while a natural alternative
is to process such components as composed entities. We propose to group such
elements in the form of quaternions and to process these quaternions using the
established quaternion algebra. Quaternion numbers and quaternion neural
networks have shown their efficiency to process multidimensional inputs as
entities, to encode internal dependencies, and to solve many tasks with less
learning parameters than real-valued models. This paper proposes to integrate
multiple feature views in quaternion-valued convolutional neural network
(QCNN), to be used for sequence-to-sequence mapping with the CTC model.
Promising results are reported using simple QCNNs in phoneme recognition
experiments with the TIMIT corpus. More precisely, QCNNs obtain a lower phoneme
error rate (PER) with less learning parameters than a competing model based on
real-valued CNNs.Comment: Accepted at INTERSPEECH 201
Leveraging native language information for improved accented speech recognition
Recognition of accented speech is a long-standing challenge for automatic
speech recognition (ASR) systems, given the increasing worldwide population of
bi-lingual speakers with English as their second language. If we consider
foreign-accented speech as an interpolation of the native language (L1) and
English (L2), using a model that can simultaneously address both languages
would perform better at the acoustic level for accented speech. In this study,
we explore how an end-to-end recurrent neural network (RNN) trained system with
English and native languages (Spanish and Indian languages) could leverage data
of native languages to improve performance for accented English speech. To this
end, we examine pre-training with native languages, as well as multi-task
learning (MTL) in which the main task is trained with native English and the
secondary task is trained with Spanish or Indian Languages. We show that the
proposed MTL model performs better than the pre-training approach and
outperforms a baseline model trained simply with English data. We suggest a new
setting for MTL in which the secondary task is trained with both English and
the native language, using the same output set. This proposed scenario yields
better performance with +11.95% and +17.55% character error rate gains over
baseline for Hispanic and Indian accents, respectively.Comment: Accepted at Interspeech 201
Context-Dependent Acoustic Modeling without Explicit Phone Clustering
Phoneme-based acoustic modeling of large vocabulary automatic speech
recognition takes advantage of phoneme context. The large number of
context-dependent (CD) phonemes and their highly varying statistics require
tying or smoothing to enable robust training. Usually, Classification and
Regression Trees are used for phonetic clustering, which is standard in Hidden
Markov Model (HMM)-based systems. However, this solution introduces a secondary
training objective and does not allow for end-to-end training. In this work, we
address a direct phonetic context modeling for the hybrid Deep Neural Network
(DNN)/HMM, that does not build on any phone clustering algorithm for the
determination of the HMM state inventory. By performing different
decompositions of the joint probability of the center phoneme state and its
left and right contexts, we obtain a factorized network consisting of different
components, trained jointly. Moreover, the representation of the phonetic
context for the network relies on phoneme embeddings. The recognition accuracy
of our proposed models on the Switchboard task is comparable and outperforms
slightly the hybrid model using the standard state-tying decision trees.Comment: Submitted to Interspeech 202
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