Authors:
Deva K Borah, Telecommunications Eng Group, RSISE, Institute of Advanced Studies, The Australian National University, Canberra ACT 0200, Australia (Australia)
Rodney A Kennedy, Telecommunications Eng Group, RSISE, Institute of Advanced Studies, The Australian National University, Canberra ACT 0200, Australia (Australia)
Inbar Fijalkow, ETIS/ENSEA - Univ. de Cergy-Pontoise, 95014 Cergy-Pontoise Cedex, France (France)
Page (NA) Paper number 1497
Abstract:
Performance of equalizers depends on the discrete time model of the
input signal and noise. The use of higher sampling rate results in
colored noise when the bandwidth of the noise-limiting prefilter is
not sufficiently large. It is shown that the mean square error (MSE)
performance of linear equalizers becomes sensitive to the decision
delay when the input noise is colored, and by using the appropriate
delay, a significant improvement in the MSE can be achieved. This is
in contrast to the behaviour observed in the white noise case well
known in the literature. This behaviour is explained in time domain
by showing the contributions to the MSE from the columns of the channel
convolution matrix and the noise eigenvectors. In the frequency domain,
it is shown that the equalizer exploits the noise correlation to improve
the MSE.
Authors:
João P Gomes,
Victor A N Barroso,
Page (NA) Paper number 1723
Abstract:
The super-exponential algorithm is a block-based technique for blind
channel equalization and system identification. Due to its fast convergence
rate, and no a priori parameterization other than the block length,
it is a useful tool for linear equalization of moderately distortive
channels. This paper presents a recursive implementation of the super-exponential
algorithm for fractionally-sampled PAM signals. Although the resulting
algorithm is still block-based, recursive propagation of several key
variables allows the block length to be significantly reduced without
compromising the algorithm's accuracy or speed, thereby enhancing its
ability to track channel variations. The convergence rate is only mildly
influenced by specific channel responses, and oversampling provides
smaller output variance and almost perfect tolerance to sampling errors.
Simulation results demonstrate the effectiveness of the proposed technique.
Authors:
Arnaud Doucet,
Christophe Andrieu,
Page (NA) Paper number 2177
Abstract:
Jump Markov linear systems (JMLS) are linear systems whose parameters
evolve with time according to a finite state Markov chain. We present
three original deterministic and stochastic iterative algorithms for
optimal state estimation of JMLS whose computational complexity at
each iteration is linear in the data length. The first algorithm yields
conditional mean estimates. The second algorithm is an algorithm that
yields the marginal maximum a posteriori (MMAP) sequence estimate of
the finite state Markov chain. The third algorithm is an algorithm
that yields the MMAP sequence estimate of the continuous state of the
JMLS. Convergence results for these three algorithms are obtained.
Computer simulations are carried out to evaluate their performance.
Authors:
Jukka Mannerkoski, Signal Processing Lab., Tampere Univ. of Technology, FINLAND (Finland)
Visa Koivunen, Signal Processing Lab., Tampere Univ. of Technology, FINLAND (Finland)
Desmond P Taylor, Dept. of Electrical and Electronic Engineering, Univ. of Canterbury, New Zealand (New Zealand)
Page (NA) Paper number 1249
Abstract:
Blind equalization of a communication channel using a prediction-based
Lattice Blind Equalizer (LBE) is considered. Second order cyclostationary
statistics and a single-input multiple-output model arising from fractional
sampling of the received data are used. The performance of the LBE
algorithm is studied in extensive simulations where commonly used example
channels are employed. Convergence in the Mean Square Error (MSE) and
Symbol Error Rate (SER) as well as the number of symbols required to
open the eye are studied at different SNRs. Robustness in the face
of channel order mismatch and channels with common subchannel zeros
is considered. The simulation results are compared to the results obtained
by the fractionally spaced Constant Modulus Algorithm, the Cyclic-RLS
algorithm and the subspace method by Moulines et al.
Authors:
Tim C Clapp,
Simon J Godsill,
Page (NA) Paper number 1701
Abstract:
We present methods for fixed-lag smoothing using Sequential Importance
sampling (SIS) on a discrete non-linear, non-Gaussian state space system
with unknown parameters. Our particular application is in the field
of digital communication systems. Each input data point is taken from
a finite set of symbols. We represent the transmission media as a fixed
filter with a finite impulse response (FIR), hence a discrete state-space
system is formed. Conventional Markov chain Monte Carlo (MCMC) techniques
such as the Gibbs sampler are unsuitable for this task because they
can only perform processing on a batch of data. Data arrives sequentially,
so it would seem sensible to process it in this way. In addition, many
communication systems are interactive, so there is a maximum level
of latency that can be tolerated before a symbol is decoded. We will
demonstrate this method by simulation and compare its performance to
existing techniques.
Authors:
Sangarapillai Lambotharan,
Jonathon A. Chambers,
Anthony G. Constantinides,
Page (NA) Paper number 1231
Abstract:
We propose a new family of mixed constant modulus algorithms for the
elimination of local minima associated with the fractionally spaced
constant modulus algorithm in the presence of channel noise. A special
case of this family is the Leaky Constant Modulus Algorithm (L-CMA).
We show that L-CMA aims to minimise jointly the intersymbol interference
(ISI) and the noise gain introduced by the equalizer. Moreover, we
derive a suitable range of leakage factors for which all local minima
due to large noise amplification are eliminated.
Authors:
Constantinos B. Papadias,
David Gesbert,
Arogyaswami J. Paulraj,
Page (NA) Paper number 2063
Abstract:
We consider the problem of linear polyphase blind equalization (BE),
i.e. we are interested in equalizing the output of a single-input-multiple-output
(SIMO) channel, without observing its input. A recent result by Liu
and showed that if the sub-channel polynomials are co-prime in the
z-domain, then the equalizer output whiteness is necessary and sufficient
for the equalization of a white input. Based on this observation, we
propose a simple decorrelation criterion for second-order based BE.
Due to its second-order nature, this criterion is insensitive to the
distance of the input from Gaussianity, hence it achieves BE even for
Gaussian or non-Gaussian inputs. Moreover, unlike other second-order
techniques, our approach bypasses channel estimation and computes directly
the equalizer. By doing so, it avoids the problem of ill-conditioning
due to channel order mismatch which is crucial to other techniques.
Combined to its good convergence properties, these characteristics
make the proposed technique an attractive option for robust polyphase
BE, as evidenced by both our analysis and computer simulation results.
Authors:
Simone Fiori,
Francesco Piazza,
Page (NA) Paper number 1911
Abstract:
The aim of this paper is to present a new cost function for blind deconvolution
of non-minimum phase systems. The proposed criterion arises as a natural
consequence of a fundamental theorem proved by Benveniste, Goursat
and Ruget, and appears to be the weighted square of the difference
among two spectra, thus its minimization leads to a weighted least-squares
blind deconvolution technique. In order to assess the new theory some
simulation results both on ideal (noiseless) and noisy channels are
presented.
Authors:
Hui Luo,
Ruey-Wen Liu,
Page (NA) Paper number 2274
Abstract:
Recent studies show that equalizers with different equalization delays
achieve different performances. The best performed equalizer is not
necessarily the one with 0-delay, and often not the case when the channel
is non-minimum-phase. In this paper, a blind channel equalization algorithm
is presented, by which equalizers with all possible equalization delays
can be calculated simultaneously from the second order statistics of
received signals. A (blind) evaluation index is then presented for
the purpose of selecting the best equalizer. Simulation shows that
the best delayed equalizer performs much better than that for other
equalization delays and those given by constant modulus algorithm and
linear prediction algorithm.
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