Abstract: The separation of speech signals has become a research
hotspot in the field of signal processing in recent years. It has
many applications and influences in teleconferencing, hearing aids,
speech recognition of machines and so on. The sounds received are
usually noisy. The issue of identifying the sounds of interest and
obtaining clear sounds in such an environment becomes a problem
worth exploring, that is, the problem of blind source separation.
This paper focuses on the under-determined blind source separation
(UBSS). Sparse component analysis is generally used for the problem
of under-determined blind source separation. The method is mainly
divided into two parts. Firstly, the clustering algorithm is used to
estimate the mixing matrix according to the observed signals. Then
the signal is separated based on the known mixing matrix. In this
paper, the problem of mixing matrix estimation is studied. This paper
proposes an improved algorithm to estimate the mixing matrix for
speech signals in the UBSS model. The traditional potential algorithm
is not accurate for the mixing matrix estimation, especially for low
signal-to noise ratio (SNR).In response to this problem, this paper
considers the idea of an improved potential function method to
estimate the mixing matrix. The algorithm not only avoids the inuence
of insufficient prior information in traditional clustering algorithm,
but also improves the estimation accuracy of mixing matrix. This
paper takes the mixing of four speech signals into two channels as
an example. The results of simulations show that the approach in this
paper not only improves the accuracy of estimation, but also applies
to any mixing matrix.
Abstract: This paper investigates MIMO (Multiple-Input
Multiple-Output) adaptive filtering techniques for the application
of supervised source separation in the context of convolutive
mixtures. From the observation that there is correlation among the
signals of the different mixtures, an improvement in the NSAF
(Normalized Subband Adaptive Filter) algorithm is proposed in
order to accelerate its convergence rate. Simulation results with
mixtures of speech signals in reverberant environments show the
superior performance of the proposed algorithm with respect to the
performances of the NLMS (Normalized Least-Mean-Square) and
conventional NSAF, considering both the convergence speed and
SIR (Signal-to-Interference Ratio) after convergence.
Abstract: This paper addresses the problem of source separation
in images. We propose a FastICA algorithm employing a modified
Gaussian contrast function for the Blind Source Separation.
Experimental result shows that the proposed Modified Gaussian
FastICA is effectively used for Blind Source Separation to obtain
better quality images. In this paper, a comparative study has been
made with other popular existing algorithms. The peak signal to
noise ratio (PSNR) and improved signal to noise ratio (ISNR) are
used as metrics for evaluating the quality of images. The ICA metric
Amari error is also used to measure the quality of separation.
Abstract: We propose a new perspective on speech
communication using blind source separation. The original speech is
mixed with key signals which consist of the mixing matrix, chaotic
signals and a random noise. However, parts of the keys (the mixing
matrix and the random noise) are not necessary in decryption. In
practice implement, one can encrypt the speech by changing the noise
signal every time. Hence, the present scheme obtains the advantages
of a One Time Pad encryption while avoiding its drawbacks in key
exchange. It is demonstrated that the proposed scheme is immune
against traditional attacks.