3,171 research outputs found
Building competitive direct acoustics-to-word models for English conversational speech recognition
Direct acoustics-to-word (A2W) models in the end-to-end paradigm have
received increasing attention compared to conventional sub-word based automatic
speech recognition models using phones, characters, or context-dependent hidden
Markov model states. This is because A2W models recognize words from speech
without any decoder, pronunciation lexicon, or externally-trained language
model, making training and decoding with such models simple. Prior work has
shown that A2W models require orders of magnitude more training data in order
to perform comparably to conventional models. Our work also showed this
accuracy gap when using the English Switchboard-Fisher data set. This paper
describes a recipe to train an A2W model that closes this gap and is at-par
with state-of-the-art sub-word based models. We achieve a word error rate of
8.8%/13.9% on the Hub5-2000 Switchboard/CallHome test sets without any decoder
or language model. We find that model initialization, training data order, and
regularization have the most impact on the A2W model performance. Next, we
present a joint word-character A2W model that learns to first spell the word
and then recognize it. This model provides a rich output to the user instead of
simple word hypotheses, making it especially useful in the case of words unseen
or rarely-seen during training.Comment: Submitted to IEEE International Conference on Acoustics, Speech and
Signal Processing (ICASSP), 201
Subword and Crossword Units for CTC Acoustic Models
This paper proposes a novel approach to create an unit set for CTC based
speech recognition systems. By using Byte Pair Encoding we learn an unit set of
an arbitrary size on a given training text. In contrast to using characters or
words as units this allows us to find a good trade-off between the size of our
unit set and the available training data. We evaluate both Crossword units,
that may span multiple word, and Subword units. By combining this approach with
decoding methods using a separate language model we are able to achieve state
of the art results for grapheme based CTC systems.Comment: Current version accepted at Interspeech 201
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
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