A maximum likelihood approach to single-channel source separation
The Journal of Machine Learning Research
Computational Auditory Scene Analysis: Principles, Algorithms, and Applications
Computational Auditory Scene Analysis: Principles, Algorithms, and Applications
General approach to blind source separation
IEEE Transactions on Signal Processing
IEEE Transactions on Signal Processing
Underdetermined blind source separation based on sparse representation
IEEE Transactions on Signal Processing
Single-Channel Speech Separation Using Soft Mask Filtering
IEEE Transactions on Audio, Speech, and Language Processing
Blind Source Separation of Postnonlinear Convolutive Mixture
IEEE Transactions on Audio, Speech, and Language Processing
IEEE Transactions on Audio, Speech, and Language Processing
A Bayesian Approach for Blind Separation of Sparse Sources
IEEE Transactions on Audio, Speech, and Language Processing
An iterative inversion approach to blind source separation
IEEE Transactions on Neural Networks
Monaural speech segregation based on pitch tracking and amplitude modulation
IEEE Transactions on Neural Networks
Nonlinear signal separation for multinonlinearity constrained mixing model
IEEE Transactions on Neural Networks
Activity index variance as an indicator of the number of signal sources
WSEAS Transactions on Signal Processing
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Blind source separation is an advanced statistical tool that has found widespread use in many signal processing applications. However, the crux topic based on one channel audio source separation has not fully developed to enable its way to laboratory implementation. The main idea approach to single channel blind source separation is based on exploiting the inherent time structure of sources known as basis filters in time domain that encode the sources in a statistically efficient manner. This paper proposes a technique for separating single channel recording of audio mixture using a hybrid of maximum likelihood and maximum a posteriori estimators. In addition, the algorithm proposes a new approach that accounts for the time structure of the speech signals by encoding them into a set of basis filters that are characteristically the most significant.