60 research outputs found
Deep neural network techniques for monaural speech enhancement: state of the art analysis
Deep neural networks (DNN) techniques have become pervasive in domains such
as natural language processing and computer vision. They have achieved great
success in these domains in task such as machine translation and image
generation. Due to their success, these data driven techniques have been
applied in audio domain. More specifically, DNN models have been applied in
speech enhancement domain to achieve denosing, dereverberation and
multi-speaker separation in monaural speech enhancement. In this paper, we
review some dominant DNN techniques being employed to achieve speech
separation. The review looks at the whole pipeline of speech enhancement from
feature extraction, how DNN based tools are modelling both global and local
features of speech and model training (supervised and unsupervised). We also
review the use of speech-enhancement pre-trained models to boost speech
enhancement process. The review is geared towards covering the dominant trends
with regards to DNN application in speech enhancement in speech obtained via a
single speaker.Comment: conferenc
TasNet: time-domain audio separation network for real-time, single-channel speech separation
Robust speech processing in multi-talker environments requires effective
speech separation. Recent deep learning systems have made significant progress
toward solving this problem, yet it remains challenging particularly in
real-time, short latency applications. Most methods attempt to construct a mask
for each source in time-frequency representation of the mixture signal which is
not necessarily an optimal representation for speech separation. In addition,
time-frequency decomposition results in inherent problems such as
phase/magnitude decoupling and long time window which is required to achieve
sufficient frequency resolution. We propose Time-domain Audio Separation
Network (TasNet) to overcome these limitations. We directly model the signal in
the time-domain using an encoder-decoder framework and perform the source
separation on nonnegative encoder outputs. This method removes the frequency
decomposition step and reduces the separation problem to estimation of source
masks on encoder outputs which is then synthesized by the decoder. Our system
outperforms the current state-of-the-art causal and noncausal speech separation
algorithms, reduces the computational cost of speech separation, and
significantly reduces the minimum required latency of the output. This makes
TasNet suitable for applications where low-power, real-time implementation is
desirable such as in hearable and telecommunication devices.Comment: Camera ready version for ICASSP 2018, Calgary, Canad
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