230 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
Direction-Aware Adaptive Online Neural Speech Enhancement with an Augmented Reality Headset in Real Noisy Conversational Environments
This paper describes the practical response- and performance-aware
development of online speech enhancement for an augmented reality (AR) headset
that helps a user understand conversations made in real noisy echoic
environments (e.g., cocktail party). One may use a state-of-the-art blind
source separation method called fast multichannel nonnegative matrix
factorization (FastMNMF) that works well in various environments thanks to its
unsupervised nature. Its heavy computational cost, however, prevents its
application to real-time processing. In contrast, a supervised beamforming
method that uses a deep neural network (DNN) for estimating spatial information
of speech and noise readily fits real-time processing, but suffers from drastic
performance degradation in mismatched conditions. Given such complementary
characteristics, we propose a dual-process robust online speech enhancement
method based on DNN-based beamforming with FastMNMF-guided adaptation. FastMNMF
(back end) is performed in a mini-batch style and the noisy and enhanced speech
pairs are used together with the original parallel training data for updating
the direction-aware DNN (front end) with backpropagation at a
computationally-allowable interval. This method is used with a blind
dereverberation method called weighted prediction error (WPE) for transcribing
the noisy reverberant speech of a speaker, which can be detected from video or
selected by a user's hand gesture or eye gaze, in a streaming manner and
spatially showing the transcriptions with an AR technique. Our experiment
showed that the word error rate was improved by more than 10 points with the
run-time adaptation using only twelve minutes of observation.Comment: IEEE/RSJ IROS 202
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Combining Localization Cues and Source Model Constraints for Binaural Source Separation
We describe a system for separating multiple sources from a two-channel recording based on interaural cues and prior knowledge of the statistics of the underlying source signals. The proposed algorithm effectively combines information derived from low level perceptual cues, similar to those used by the human auditory system, with higher level information related to speaker identity. We combine a probabilistic model of the observed interaural level and phase differences with a prior model of the source statistics and derive an EM algorithm for finding the maximum likelihood parameters of the joint model. The system is able to separate more sound sources than there are observed channels in the presence of reverberation. In simulated mixtures of speech from two and three speakers the proposed algorithm gives a signal-to-noise ratio improvement of 1.7 dB over a baseline algorithm which uses only interaural cues. Further improvement is obtained by incorporating eigenvoice speaker adaptation to enable the source model to better match the sources present in the signal. This improves performance over the baseline by 2.7 dB when the speakers used for training and testing are matched. However, the improvement is minimal when the test data is very different from that used in training
SD-TEAM: Interactive Learning, Self-Evaluation and Multimodal Technologies for Multidomain Spoken Dialog Systems
Speech technology currently supports the development of dialogue systems that function in limited domains for which they were trained and in conditions for which they were designed, that is, specific acoustic conditions, speakers etc. The international scientific community has made significant efforts in exploring methods for adaptation to different acoustic contexts, tasks and types of user. However, further work is needed to produce multimodal spoken dialogue systems capable of exploiting interactivity to learn online in order to improve their performance. The goal is to produce flexible and dynamic multimodal, interactive systems based on spoken communication, capable of detecting automatically their operating conditions and especially of learning from user interactions and experience through evaluating their own performance. Such ?living? systems will evolve continuously and without supervision until user satisfaction is achieved. Special attention will be paid to those groups of users for which adaptation and personalisation is essential: amongst others, people with disabilities which lead to communication difficulties (hearing loss, dysfluent speech, ...), mobility problems and non-native users. In this context, the SD-TEAM Project aims to advance the development of technologies for interactive learning and evaluation. In addition, it will develop flexible distributed architectures that allow synergistic interaction between processing modules from a variety of dialogue systems designed for distinct tasks, user groups, acoustic conditions, etc. These technologies will be demonstrated via multimodal dialogue systems to access to services from home and to access to unstructured information, based on the multi-domain systems developed in the previous project TIN2005-08660-C04
Studies on noise robust automatic speech recognition
Noise in everyday acoustic environments such as cars, traffic environments, and cafeterias remains one of the main challenges in automatic speech recognition (ASR). As a research theme, it has received wide attention in conferences and scientific journals focused on speech technology. This article collection reviews both the classic and novel approaches suggested for noise robust ASR. The articles are literature reviews written for the spring 2009 seminar course on noise robust automatic speech recognition (course code T-61.6060) held at TKK
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