146,164 research outputs found
Shift-Invariant Kernel Additive Modelling for Audio Source Separation
A major goal in blind source separation to identify and separate sources is
to model their inherent characteristics. While most state-of-the-art approaches
are supervised methods trained on large datasets, interest in non-data-driven
approaches such as Kernel Additive Modelling (KAM) remains high due to their
interpretability and adaptability. KAM performs the separation of a given
source applying robust statistics on the time-frequency bins selected by a
source-specific kernel function, commonly the K-NN function. This choice
assumes that the source of interest repeats in both time and frequency. In
practice, this assumption does not always hold. Therefore, we introduce a
shift-invariant kernel function capable of identifying similar spectral content
even under frequency shifts. This way, we can considerably increase the amount
of suitable sound material available to the robust statistics. While this leads
to an increase in separation performance, a basic formulation, however, is
computationally expensive. Therefore, we additionally present acceleration
techniques that lower the overall computational complexity.Comment: Feedback is welcom
Adaptation of speaker-specific bases in non-negative matrix factorization for single channel speech-music separation
This paper introduces a speaker adaptation algorithm for nonnegative matrix factorization (NMF) models. The proposed adaptation algorithm is a combination of Bayesian and subspace model adaptation. The adapted model is used to separate speech signal from a background music signal in a single record. Training speech data for multiple speakers is used with NMF to train a set of basis vectors as a general model for speech signals. The probabilistic interpretation of NMF is used to achieve Bayesian adaptation to adjust the general model with respect to the actual properties of the speech signals that is observed in the mixed signal. The Bayesian adapted model is adapted again by a linear transform, which changes the subspace that the Bayesian adapted model spans to better match the speech signal that is in the mixed signal. The experimental results show that combining Bayesian with linear transform adaptation improves the separation results
Semi-Supervised Sound Source Localization Based on Manifold Regularization
Conventional speaker localization algorithms, based merely on the received
microphone signals, are often sensitive to adverse conditions, such as: high
reverberation or low signal to noise ratio (SNR). In some scenarios, e.g. in
meeting rooms or cars, it can be assumed that the source position is confined
to a predefined area, and the acoustic parameters of the environment are
approximately fixed. Such scenarios give rise to the assumption that the
acoustic samples from the region of interest have a distinct geometrical
structure. In this paper, we show that the high dimensional acoustic samples
indeed lie on a low dimensional manifold and can be embedded into a low
dimensional space. Motivated by this result, we propose a semi-supervised
source localization algorithm which recovers the inverse mapping between the
acoustic samples and their corresponding locations. The idea is to use an
optimization framework based on manifold regularization, that involves
smoothness constraints of possible solutions with respect to the manifold. The
proposed algorithm, termed Manifold Regularization for Localization (MRL), is
implemented in an adaptive manner. The initialization is conducted with only
few labelled samples attached with their respective source locations, and then
the system is gradually adapted as new unlabelled samples (with unknown source
locations) are received. Experimental results show superior localization
performance when compared with a recently presented algorithm based on a
manifold learning approach and with the generalized cross-correlation (GCC)
algorithm as a baseline
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