13,404 research outputs found
On Training Targets and Objective Functions for Deep-Learning-Based Audio-Visual Speech Enhancement
Audio-visual speech enhancement (AV-SE) is the task of improving speech
quality and intelligibility in a noisy environment using audio and visual
information from a talker. Recently, deep learning techniques have been adopted
to solve the AV-SE task in a supervised manner. In this context, the choice of
the target, i.e. the quantity to be estimated, and the objective function,
which quantifies the quality of this estimate, to be used for training is
critical for the performance. This work is the first that presents an
experimental study of a range of different targets and objective functions used
to train a deep-learning-based AV-SE system. The results show that the
approaches that directly estimate a mask perform the best overall in terms of
estimated speech quality and intelligibility, although the model that directly
estimates the log magnitude spectrum performs as good in terms of estimated
speech quality
An Overview of Deep-Learning-Based Audio-Visual Speech Enhancement and Separation
Speech enhancement and speech separation are two related tasks, whose purpose
is to extract either one or more target speech signals, respectively, from a
mixture of sounds generated by several sources. Traditionally, these tasks have
been tackled using signal processing and machine learning techniques applied to
the available acoustic signals. Since the visual aspect of speech is
essentially unaffected by the acoustic environment, visual information from the
target speakers, such as lip movements and facial expressions, has also been
used for speech enhancement and speech separation systems. In order to
efficiently fuse acoustic and visual information, researchers have exploited
the flexibility of data-driven approaches, specifically deep learning,
achieving strong performance. The ceaseless proposal of a large number of
techniques to extract features and fuse multimodal information has highlighted
the need for an overview that comprehensively describes and discusses
audio-visual speech enhancement and separation based on deep learning. In this
paper, we provide a systematic survey of this research topic, focusing on the
main elements that characterise the systems in the literature: acoustic
features; visual features; deep learning methods; fusion techniques; training
targets and objective functions. In addition, we review deep-learning-based
methods for speech reconstruction from silent videos and audio-visual sound
source separation for non-speech signals, since these methods can be more or
less directly applied to audio-visual speech enhancement and separation.
Finally, we survey commonly employed audio-visual speech datasets, given their
central role in the development of data-driven approaches, and evaluation
methods, because they are generally used to compare different systems and
determine their performance
Effects of Lombard Reflex on the Performance of Deep-Learning-Based Audio-Visual Speech Enhancement Systems
Humans tend to change their way of speaking when they are immersed in a noisy
environment, a reflex known as Lombard effect. Current speech enhancement
systems based on deep learning do not usually take into account this change in
the speaking style, because they are trained with neutral (non-Lombard) speech
utterances recorded under quiet conditions to which noise is artificially
added. In this paper, we investigate the effects that the Lombard reflex has on
the performance of audio-visual speech enhancement systems based on deep
learning. The results show that a gap in the performance of as much as
approximately 5 dB between the systems trained on neutral speech and the ones
trained on Lombard speech exists. This indicates the benefit of taking into
account the mismatch between neutral and Lombard speech in the design of
audio-visual speech enhancement systems
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
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