62 research outputs found
English Broadcast News Speech Recognition by Humans and Machines
With recent advances in deep learning, considerable attention has been given
to achieving automatic speech recognition performance close to human
performance on tasks like conversational telephone speech (CTS) recognition. In
this paper we evaluate the usefulness of these proposed techniques on broadcast
news (BN), a similar challenging task. We also perform a set of recognition
measurements to understand how close the achieved automatic speech recognition
results are to human performance on this task. On two publicly available BN
test sets, DEV04F and RT04, our speech recognition system using LSTM and
residual network based acoustic models with a combination of n-gram and neural
network language models performs at 6.5% and 5.9% word error rate. By achieving
new performance milestones on these test sets, our experiments show that
techniques developed on other related tasks, like CTS, can be transferred to
achieve similar performance. In contrast, the best measured human recognition
performance on these test sets is much lower, at 3.6% and 2.8% respectively,
indicating that there is still room for new techniques and improvements in this
space, to reach human performance levels.Comment: \copyright 2019 IEEE. Personal use of this material is permitted.
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this work in other work
Relative Positional Encoding for Speech Recognition and Direct Translation
Transformer models are powerful sequence-to-sequence architectures that are
capable of directly mapping speech inputs to transcriptions or translations.
However, the mechanism for modeling positions in this model was tailored for
text modeling, and thus is less ideal for acoustic inputs. In this work, we
adapt the relative position encoding scheme to the Speech Transformer, where
the key addition is relative distance between input states in the
self-attention network. As a result, the network can better adapt to the
variable distributions present in speech data. Our experiments show that our
resulting model achieves the best recognition result on the Switchboard
benchmark in the non-augmentation condition, and the best published result in
the MuST-C speech translation benchmark. We also show that this model is able
to better utilize synthetic data than the Transformer, and adapts better to
variable sentence segmentation quality for speech translation.Comment: Submitted to Interspeech 202
On the Choice of Modeling Unit for Sequence-to-Sequence Speech Recognition
In conventional speech recognition, phoneme-based models outperform
grapheme-based models for non-phonetic languages such as English. The
performance gap between the two typically reduces as the amount of training
data is increased. In this work, we examine the impact of the choice of
modeling unit for attention-based encoder-decoder models. We conduct
experiments on the LibriSpeech 100hr, 460hr, and 960hr tasks, using various
target units (phoneme, grapheme, and word-piece); across all tasks, we find
that grapheme or word-piece models consistently outperform phoneme-based
models, even though they are evaluated without a lexicon or an external
language model. We also investigate model complementarity: we find that we can
improve WERs by up to 9% relative by rescoring N-best lists generated from a
strong word-piece based baseline with either the phoneme or the grapheme model.
Rescoring an N-best list generated by the phonemic system, however, provides
limited improvements. Further analysis shows that the word-piece-based models
produce more diverse N-best hypotheses, and thus lower oracle WERs, than
phonemic models.Comment: To appear in the proceedings of INTERSPEECH 201
RWTH ASR Systems for LibriSpeech: Hybrid vs Attention -- w/o Data Augmentation
We present state-of-the-art automatic speech recognition (ASR) systems
employing a standard hybrid DNN/HMM architecture compared to an attention-based
encoder-decoder design for the LibriSpeech task. Detailed descriptions of the
system development, including model design, pretraining schemes, training
schedules, and optimization approaches are provided for both system
architectures. Both hybrid DNN/HMM and attention-based systems employ
bi-directional LSTMs for acoustic modeling/encoding. For language modeling, we
employ both LSTM and Transformer based architectures. All our systems are built
using RWTHs open-source toolkits RASR and RETURNN. To the best knowledge of the
authors, the results obtained when training on the full LibriSpeech training
set, are the best published currently, both for the hybrid DNN/HMM and the
attention-based systems. Our single hybrid system even outperforms previous
results obtained from combining eight single systems. Our comparison shows that
on the LibriSpeech 960h task, the hybrid DNN/HMM system outperforms the
attention-based system by 15% relative on the clean and 40% relative on the
other test sets in terms of word error rate. Moreover, experiments on a reduced
100h-subset of the LibriSpeech training corpus even show a more pronounced
margin between the hybrid DNN/HMM and attention-based architectures.Comment: Proceedings of INTERSPEECH 201
Improving sequence-to-sequence speech recognition training with on-the-fly data augmentation
Sequence-to-Sequence (S2S) models recently started to show state-of-the-art
performance for automatic speech recognition (ASR). With these large and deep
models overfitting remains the largest problem, outweighing performance
improvements that can be obtained from better architectures. One solution to
the overfitting problem is increasing the amount of available training data and
the variety exhibited by the training data with the help of data augmentation.
In this paper we examine the influence of three data augmentation methods on
the performance of two S2S model architectures. One of the data augmentation
method comes from literature, while two other methods are our own development -
a time perturbation in the frequency domain and sub-sequence sampling. Our
experiments on Switchboard and Fisher data show state-of-the-art performance
for S2S models that are trained solely on the speech training data and do not
use additional text data.Comment: To appear in ICASSP 202
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