23 research outputs found
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Confidence Estimation for Black Box Automatic Speech Recognition Systems Using Lattice Recurrent Neural Networks
System-independent ASR error detection and classification using Recurrent Neural Network
This paper addresses errors in continuous Automatic Speech Recognition (ASR) in two stages: error detection and error type classification. Unlike the majority of research in this field, we propose to handle the recognition errors independently from the ASR decoder. We first establish an effective set of generic features derived exclusively from the recognizer output to compensate for the absence of ASR decoder information. Then, we apply a variant Recurrent Neural Network (V-RNN) based models for error detection and error type classification. Such model learn additional information to the recognized word classification using label dependency. As a result, experiments on Multi-Genre Broadcast Media corpus have shown that the proposed generic features setup leads to achieve competitive performances, compared to state of the art systems in both tasks. Furthermore, we have shown that V-RNN trained on the proposed feature set appear to be an effective classifier for the ASR error detection with an Accuracy of 85.43%
Détection et correction d'erreurs utilisant les probabilités a posteriori dans un système de reconnaissance de phrases manuscrites en-ligne
National audienceDans cet article, nous présentons un système complet de reconnaissance de phrases manuscrites en-ligne. Nous nous intéressons plus particulièrement à la détection d'erreurs potentielles sur les phrases issues d'une reconnaissance avec une approche au Maximum A Posteriori. Les probabilités a posteriori des mots, obtenues à partir d'une représentation sous la forme d'un réseau de confusion, sont ainsi utilisées comme indices de confiance. Des classifieurs dédiés (ici, des SVM) sont ensuite appris afin de corriger ces erreurs, en combinant ces probabilités a posteriori à d'autres sources de connaissance. Un mécanisme de rejet est également introduit afin de distinguer les hypothèses d'erreur qui ne pourront être corrigées par l'approche proposée. Des expérimentations ont été menées sur une base de 425 phrases manuscrites écrites par 17 scripteurs. Elles ont mis en évidence une réduction relative du taux d'erreur sur les mots de 14,6
Semi-Supervised Acoustic Model Training by Discriminative Data Selection from Multiple ASR Systems' Hypotheses
While the performance of ASR systems depends on the size of the training data, it is very costly to prepare accurate and faithful transcripts. In this paper, we investigate a semisupervised training scheme, which takes the advantage of huge quantities of unlabeled video lecture archive, particularly for the deep neural network (DNN) acoustic model. In the proposed method, we obtain ASR hypotheses by complementary GMM-and DNN-based ASR systems. Then, a set of CRF-based classifiers is trained to select the correct hypotheses and verify the selected data. The proposed hypothesis combination shows higher quality compared with the conventional system combination method (ROVER). Moreover, compared with the conventional data selection based on confidence measure score, our method is demonstrated more effective for filtering usable data. Significant improvement in the ASR accuracy is achieved over the baseline system and in comparison with the models trained with the conventional system combination and data selection methods
Deep neural network features and semi-supervised training for low resource speech recognition
We propose a new technique for training deep neural networks (DNNs) as data-driven feature front-ends for large vocabulary con-tinuous speech recognition (LVCSR) in low resource settings. To circumvent the lack of sufficient training data for acoustic mod-eling in these scenarios, we use transcribed multilingual data and semi-supervised training to build the proposed feature front-ends. In our experiments, the proposed features provide an absolute im-provement of 16 % in a low-resource LVCSR setting with only one hour of in-domain training data. While close to three-fourths of these gains come from DNN-based features, the remaining are from semi-supervised training. Index Terms — Low resource, speech recognition, deep neural networks, semi-supervised training, bottleneck features
Confidence Score Based Speaker Adaptation of Conformer Speech Recognition Systems
Speaker adaptation techniques provide a powerful solution to customise
automatic speech recognition (ASR) systems for individual users. Practical
application of unsupervised model-based speaker adaptation techniques to data
intensive end-to-end ASR systems is hindered by the scarcity of speaker-level
data and performance sensitivity to transcription errors. To address these
issues, a set of compact and data efficient speaker-dependent (SD) parameter
representations are used to facilitate both speaker adaptive training and
test-time unsupervised speaker adaptation of state-of-the-art Conformer ASR
systems. The sensitivity to supervision quality is reduced using a confidence
score-based selection of the less erroneous subset of speaker-level adaptation
data. Two lightweight confidence score estimation modules are proposed to
produce more reliable confidence scores. The data sparsity issue, which is
exacerbated by data selection, is addressed by modelling the SD parameter
uncertainty using Bayesian learning. Experiments on the benchmark 300-hour
Switchboard and the 233-hour AMI datasets suggest that the proposed confidence
score-based adaptation schemes consistently outperformed the baseline
speaker-independent (SI) Conformer model and conventional non-Bayesian, point
estimate-based adaptation using no speaker data selection. Similar consistent
performance improvements were retained after external Transformer and LSTM
language model rescoring. In particular, on the 300-hour Switchboard corpus,
statistically significant WER reductions of 1.0%, 1.3%, and 1.4% absolute
(9.5%, 10.9%, and 11.3% relative) were obtained over the baseline SI Conformer
on the NIST Hub5'00, RT02, and RT03 evaluation sets respectively. Similar WER
reductions of 2.7% and 3.3% absolute (8.9% and 10.2% relative) were also
obtained on the AMI development and evaluation sets.Comment: IEEE/ACM Transactions on Audio, Speech, and Language Processin