434 research outputs found
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
Light Gated Recurrent Units for Speech Recognition
A field that has directly benefited from the recent advances in deep learning
is Automatic Speech Recognition (ASR). Despite the great achievements of the
past decades, however, a natural and robust human-machine speech interaction
still appears to be out of reach, especially in challenging environments
characterized by significant noise and reverberation. To improve robustness,
modern speech recognizers often employ acoustic models based on Recurrent
Neural Networks (RNNs), that are naturally able to exploit large time contexts
and long-term speech modulations. It is thus of great interest to continue the
study of proper techniques for improving the effectiveness of RNNs in
processing speech signals.
In this paper, we revise one of the most popular RNN models, namely Gated
Recurrent Units (GRUs), and propose a simplified architecture that turned out
to be very effective for ASR. The contribution of this work is two-fold: First,
we analyze the role played by the reset gate, showing that a significant
redundancy with the update gate occurs. As a result, we propose to remove the
former from the GRU design, leading to a more efficient and compact single-gate
model. Second, we propose to replace hyperbolic tangent with ReLU activations.
This variation couples well with batch normalization and could help the model
learn long-term dependencies without numerical issues.
Results show that the proposed architecture, called Light GRU (Li-GRU), not
only reduces the per-epoch training time by more than 30% over a standard GRU,
but also consistently improves the recognition accuracy across different tasks,
input features, noisy conditions, as well as across different ASR paradigms,
ranging from standard DNN-HMM speech recognizers to end-to-end CTC models.Comment: Copyright 2018 IEE
PHONOTACTIC AND ACOUSTIC LANGUAGE RECOGNITION
Práce pojednává o fonotaktickĂ©m a akustickĂ©m pĹ™Ăstupu pro automatickĂ© rozpoznávánĂ jazyka. Prvnà část práce pojednává o fonotaktickĂ©m pĹ™Ăstupu zaloĹľenĂ©m na vĂ˝skytu fonĂ©movĂ˝ch sekvenci v Ĺ™eÄŤi. NejdĹ™Ăve je prezentován popis vĂ˝voje fonĂ©movĂ©ho rozpoznávaÄŤe jako techniky pro pĹ™epis Ĺ™eÄŤi do sekvence smysluplnĂ˝ch symbolĹŻ. HlavnĂ dĹŻraz je kladen na dobrĂ© natrĂ©novánĂ fonĂ©movĂ©ho rozpoznávaÄŤe a kombinaci vĂ˝sledkĹŻ z nÄ›kolika fonĂ©movĂ˝ch rozpoznávaÄŤĹŻ trĂ©novanĂ˝ch na rĹŻznĂ˝ch jazycĂch (ParalelnĂ fonĂ©movĂ© rozpoznávánĂ následovanĂ© jazykovĂ˝mi modely (PPRLM)). Práce takĂ© pojednává o novĂ© technice anti-modely v PPRLM a studuje pouĹľitĂ fonĂ©movĂ˝ch grafĹŻ mĂsto nejlepšĂho pĹ™episu. Na závÄ›r práce jsou porovnány dva pĹ™Ăstupy modelovánĂ vĂ˝stupu fonĂ©movĂ©ho rozpoznávaÄŤe -- standardnĂ n-gramovĂ© jazykovĂ© modely a binárnĂ rozhodovacĂ stromy. HlavnĂ pĹ™Ănos v akustickĂ©m pĹ™Ăstupu je diskriminativnĂ modelovánĂ cĂlovĂ˝ch modelĹŻ jazykĹŻ a prvnĂ experimenty s kombinacĂ diskriminativnĂho trĂ©novánĂ a na pĹ™ĂznacĂch, kde byl odstranÄ›n vliv kanálu. Práce dále zkoumá rĹŻznĂ© druhy technik fĂşzi akustickĂ©ho a fonotaktickĂ©ho pĹ™Ăstupu. Všechny experimenty jsou provedeny na standardnĂch datech z NIST evaluaci konanĂ© v letech 2003, 2005 a 2007, takĹľe jsou pĹ™Ămo porovnatelnĂ© s vĂ˝sledky ostatnĂch skupin zabĂ˝vajĂcĂch se automatickĂ˝m rozpoznávánĂm jazyka. S fĂşzĂ uvedenĂ˝ch technik jsme posunuli state-of-the-art vĂ˝sledky a dosáhli vynikajĂcĂch vĂ˝sledkĹŻ ve dvou NIST evaluacĂch.This thesis deals with phonotactic and acoustic techniques for automatic language recognition (LRE). The first part of the thesis deals with the phonotactic language recognition based on co-occurrences of phone sequences in speech. A thorough study of phone recognition as tokenization technique for LRE is done, with focus on the amounts of training data for phone recognizer and on the combination of phone recognizers trained on several language (Parallel Phone Recognition followed by Language Model - PPRLM). The thesis also deals with novel technique of anti-models in PPRLM and investigates into using phone lattices instead of strings. The work on phonotactic approach is concluded by a comparison of classical n-gram modeling techniques and binary decision trees. The acoustic LRE was addressed too, with the main focus on discriminative techniques for training target language acoustic models and on initial (but successful) experiments with removing channel dependencies. We have also investigated into the fusion of phonotactic and acoustic approaches. All experiments were performed on standard data from NIST 2003, 2005 and 2007 evaluations so that the results are directly comparable to other laboratories in the LRE community. With the above mentioned techniques, the fused systems defined the state-of-the-art in the LRE field and reached excellent results in NIST evaluations.
An on-line speaker adaptation method for HMM-based speech recognizers
In the past few years numerous techniques have been proposed to improve the efficiency of basic adaptation methods like MLLR and MAP. These adaptation methods have a common aim, which is to increase the likelihood of the phoneme models for a particular speaker. During their operation, these speaker adaptation methods need precise phonetic segmentation information of the actual utterance, but these data samples are often faulty. To improve the overall performance, only those frames from the spoken sentence which are well segmented should be retained, while the incorrectly segmented data should not be used during adaptation. Several heuristic algorithms have been proposed in the literature for the selection of the reliably segmented data blocks, and here we would like to suggest some new heuristics that discriminate between faulty and well-segmented data. The effect of these methods on the efficiency of speech recognition using speaker adaptation is examined, and conclusions for each will be drawn. Besided post-filtering the set of the segmented adaptation examples, another way of improving the efficiency of the adaptation method might be to create a more precise segmentation, which should then reduce the chance of faulty data samples being included. We suggest a method like this here as well which is based on a scoring procedure for the N-best lists, taking into account phoneme duration
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