4,089 research outputs found
Deep Learning for Environmentally Robust Speech Recognition: An Overview of Recent Developments
Eliminating the negative effect of non-stationary environmental noise is a
long-standing research topic for automatic speech recognition that stills
remains an important challenge. Data-driven supervised approaches, including
ones based on deep neural networks, have recently emerged as potential
alternatives to traditional unsupervised approaches and with sufficient
training, can alleviate the shortcomings of the unsupervised methods in various
real-life acoustic environments. In this light, we review recently developed,
representative deep learning approaches for tackling non-stationary additive
and convolutional degradation of speech with the aim of providing guidelines
for those involved in the development of environmentally robust speech
recognition systems. We separately discuss single- and multi-channel techniques
developed for the front-end and back-end of speech recognition systems, as well
as joint front-end and back-end training frameworks
DNN-Based Multi-Frame MVDR Filtering for Single-Microphone Speech Enhancement
Multi-frame approaches for single-microphone speech enhancement, e.g., the
multi-frame minimum-variance-distortionless-response (MVDR) filter, are able to
exploit speech correlations across neighboring time frames. In contrast to
single-frame approaches such as the Wiener gain, it has been shown that
multi-frame approaches achieve a substantial noise reduction with hardly any
speech distortion, provided that an accurate estimate of the correlation
matrices and especially the speech interframe correlation vector is available.
Typical estimation procedures of the correlation matrices and the speech
interframe correlation (IFC) vector require an estimate of the speech presence
probability (SPP) in each time-frequency bin. In this paper, we propose to use
a bi-directional long short-term memory deep neural network (DNN) to estimate a
speech mask and a noise mask for each time-frequency bin, using which two
different SPP estimates are derived. Aiming at achieving a robust performance,
the DNN is trained for various noise types and signal-to-noise ratios.
Experimental results show that the multi-frame MVDR in combination with the
proposed data-driven SPP estimator yields an increased speech quality compared
to a state-of-the-art model-based estimator
Multi-talker Speech Separation with Utterance-level Permutation Invariant Training of Deep Recurrent Neural Networks
In this paper we propose the utterance-level Permutation Invariant Training
(uPIT) technique. uPIT is a practically applicable, end-to-end, deep learning
based solution for speaker independent multi-talker speech separation.
Specifically, uPIT extends the recently proposed Permutation Invariant Training
(PIT) technique with an utterance-level cost function, hence eliminating the
need for solving an additional permutation problem during inference, which is
otherwise required by frame-level PIT. We achieve this using Recurrent Neural
Networks (RNNs) that, during training, minimize the utterance-level separation
error, hence forcing separated frames belonging to the same speaker to be
aligned to the same output stream. In practice, this allows RNNs, trained with
uPIT, to separate multi-talker mixed speech without any prior knowledge of
signal duration, number of speakers, speaker identity or gender. We evaluated
uPIT on the WSJ0 and Danish two- and three-talker mixed-speech separation tasks
and found that uPIT outperforms techniques based on Non-negative Matrix
Factorization (NMF) and Computational Auditory Scene Analysis (CASA), and
compares favorably with Deep Clustering (DPCL) and the Deep Attractor Network
(DANet). Furthermore, we found that models trained with uPIT generalize well to
unseen speakers and languages. Finally, we found that a single model, trained
with uPIT, can handle both two-speaker, and three-speaker speech mixtures
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