673 research outputs found
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
Self-Attention Networks for Connectionist Temporal Classification in Speech Recognition
The success of self-attention in NLP has led to recent applications in
end-to-end encoder-decoder architectures for speech recognition. Separately,
connectionist temporal classification (CTC) has matured as an alignment-free,
non-autoregressive approach to sequence transduction, either by itself or in
various multitask and decoding frameworks. We propose SAN-CTC, a deep, fully
self-attentional network for CTC, and show it is tractable and competitive for
end-to-end speech recognition. SAN-CTC trains quickly and outperforms existing
CTC models and most encoder-decoder models, with character error rates (CERs)
of 4.7% in 1 day on WSJ eval92 and 2.8% in 1 week on LibriSpeech test-clean,
with a fixed architecture and one GPU. Similar improvements hold for WERs after
LM decoding. We motivate the architecture for speech, evaluate position and
downsampling approaches, and explore how label alphabets (character, phoneme,
subword) affect attention heads and performance.Comment: Accepted to ICASSP 201
A hierarchy of recurrent networks for speech recognition
Generative models for sequential data based on directed graphs of Restricted Boltzmann Machines (RBMs) are able to accurately model high dimensional sequences as recently shown. In these models, temporal dependencies in the input are discovered by either buffering previous visible variables or by recurrent connections of the hidden variables. Here we propose a modification of these models, the Temporal Reservoir Machine (TRM). It utilizes a recurrent artificial neural network (ANN) for integrating information from the input over
time. This information is then fed into a RBM at each time step. To avoid difficulties of recurrent network learning, the ANN remains untrained and hence can be thought of as a random feature extractor. Using the architecture of multi-layer RBMs (Deep Belief Networks), the TRMs can be used as a building block for complex hierarchical models. This approach unifies RBM-based approaches for sequential data modeling and the Echo State Network, a powerful approach for black-box system identification. The TRM is tested on a spoken digits task under noisy conditions, and competitive performances compared to previous models are observed
Neural Speech Synthesis with Transformer Network
Although end-to-end neural text-to-speech (TTS) methods (such as Tacotron2)
are proposed and achieve state-of-the-art performance, they still suffer from
two problems: 1) low efficiency during training and inference; 2) hard to model
long dependency using current recurrent neural networks (RNNs). Inspired by the
success of Transformer network in neural machine translation (NMT), in this
paper, we introduce and adapt the multi-head attention mechanism to replace the
RNN structures and also the original attention mechanism in Tacotron2. With the
help of multi-head self-attention, the hidden states in the encoder and decoder
are constructed in parallel, which improves the training efficiency. Meanwhile,
any two inputs at different times are connected directly by self-attention
mechanism, which solves the long range dependency problem effectively. Using
phoneme sequences as input, our Transformer TTS network generates mel
spectrograms, followed by a WaveNet vocoder to output the final audio results.
Experiments are conducted to test the efficiency and performance of our new
network. For the efficiency, our Transformer TTS network can speed up the
training about 4.25 times faster compared with Tacotron2. For the performance,
rigorous human tests show that our proposed model achieves state-of-the-art
performance (outperforms Tacotron2 with a gap of 0.048) and is very close to
human quality (4.39 vs 4.44 in MOS)
OverFlow: Putting flows on top of neural transducers for better TTS
Neural HMMs are a type of neural transducer recently proposed for
sequence-to-sequence modelling in text-to-speech. They combine the best
features of classic statistical speech synthesis and modern neural TTS,
requiring less data and fewer training updates, and are less prone to gibberish
output caused by neural attention failures. In this paper, we combine neural
HMM TTS with normalising flows for describing the highly non-Gaussian
distribution of speech acoustics. The result is a powerful, fully probabilistic
model of durations and acoustics that can be trained using exact maximum
likelihood. Experiments show that a system based on our proposal needs fewer
updates than comparable methods to produce accurate pronunciations and a
subjective speech quality close to natural speech. Please see
https://shivammehta25.github.io/OverFlow/ for audio examples and code.Comment: 5 pages, 2 figures. Accepted for publication at Interspeech 202
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