131 research outputs found
Wireless Channel Equalization in Digital Communication Systems
Our modern society has transformed to an information-demanding system, seeking voice, video, and data in quantities that could not be imagined even a decade ago. The mobility of communicators has added more challenges. One of the new challenges is to conceive highly reliable and fast communication system unaffected by the problems caused in the multipath fading wireless channels. Our quest is to remove one of the obstacles in the way of achieving ultimately fast and reliable wireless digital communication, namely Inter-Symbol Interference (ISI), the intensity of which makes the channel noise inconsequential.
The theoretical background for wireless channels modeling and adaptive signal processing are covered in first two chapters of dissertation.
The approach of this thesis is not based on one methodology but several algorithms and configurations that are proposed and examined to fight the ISI problem. There are two main categories of channel equalization techniques, supervised (training) and blind unsupervised (blind) modes. We have studied the application of a new and specially modified neural network requiring very short training period for the proper channel equalization in supervised mode. The promising performance in the graphs for this network is presented in chapter 4.
For blind modes two distinctive methodologies are presented and studied. Chapter 3 covers the concept of multiple cooperative algorithms for the cases of two and three cooperative algorithms. The select absolutely larger equalized signal and majority vote methods have been used in 2-and 3-algoirithm systems respectively. Many of the demonstrated results are encouraging for further research.
Chapter 5 involves the application of general concept of simulated annealing in blind mode equalization. A limited strategy of constant annealing noise is experimented for testing the simple algorithms used in multiple systems. Convergence to local stationary points of the cost function in parameter space is clearly demonstrated and that justifies the use of additional noise. The capability of the adding the random noise to release the algorithm from the local traps is established in several cases
Performance Evaluation of Adaptive Equalizer in a Communication System
This project deals with the study of the various kinds of interferences in a communication channel viz. Inter symbol Interference, Multipath Interference and Additive Interference. It deals with the design of an Adaptive Equalizer. The idea of the equalizer is to build (another) filter in the receiver that counteracts the effect of the channel. In essence, the equalizer must “unscatter” the impulse response. This can be stated as the goal of designing the equalizer E so that the impulse response of the combined channel and equalizer CE has a single spike. This can be solved using different techniques.
In this project, we have implemented an ‘Adaptive Equalizer’ using four different algorithms in Matlab. We have suggested different ways to decide the coefficients of the equalizer. The first procedure (LEAST SQUARE ALGORITHM) minimizes the square of the symbol recovery error over a block of data which can be done by using matrix pseudo inversion. The second method (LEAST MEAN SQUARE ALGORITHM) involves minimizing the square of the error between the received data values and the transmitted values which are achieved via an adaptive element. The third method (DECISION DIRECTED ALGORITHM) and the fourth method (DISPERSION MINIMIZING ALGORITHM) are used when there is no training sequence and other performance functions are appropriate.
In addition to this we have undertaken a study and realization of the Bit Error Rate of a communication system using VisSim Software
Recommended from our members
Channel equalization to achieve high bit rates in discrete multitone systems
textMulticarrier modulation (MCM) techniques such as orthogonal frequency division
multiplexing (OFDM) and discrete multi-tone (DMT) modulation are attractive
for high-speed data communications due to the ease with which MCM can combat
channel dispersion. With all the benefits MCM could give, DMT modulation has an
extra ability to perform dynamic bit loading, which has the potential to exploit fully
the available bandwidth in a slowly time-varying channel. In broadband wireline
communications, DMT modulation is standardized for asymmetric digital subscribe
line (ADSL) and very-high-bit-rate digital subscriber line (VDSL) modems. ADSL
and VDSL standards are used by telephone companies to provide high speed data
service to residences and offices.
In an ADSL receiver, an equalizer is required to compensate for the channel’s
dispersion in the time domain and the channel’s distortion in the frequency domain
of the transmitted waveform. This dissertation proposes design methods for linear
equalizers to increase the bit rate of the connection. The methods are amenable
to implementation on programmable fixed-point digital signal processors, which are
employed in ADSL/VDSL transceivers.
A conventional ADSL equalizer consists of a time-domain equalizer, a fast
Fourier transform, and a frequency domain equalizer. The time domain equalizer
(TEQ) is a finite impulse response filter that when coupled with a discretized channel
produces an equivalent channel whose impulse response is shorter than that of
the discretized channel. This channel shortening is required by the ADSL standards.
In this dissertation, I first propose a linear phase TEQ design that exploits symmetry
in existing eigen-filter approaches such as minimum mean square error(MMSE),
maximum shortening signal to noise ratio (MSSNR) and minimum intersymbol interference
(Min-ISI) equalizers. TEQs with symmetric coefficients can reach the
same performance as non-symmetric ones with much lower training complexity.
Second, I improve Min-ISI design. I reformulate the cost function to make
long TEQs design feasible. I remove the dependency of transmission delay in order
to reduce the complexity associated with delay optimization. The quantized
weighting is introduced to further lower the complexity. I also propose an iterative
optimization procedure of Min-ISI that completely avoids Cholesky decomposition
hence is better suited for a fixed-point implementation.
Finally I propose a dual-path TEQ structure, which designs a standard singleFIR
TEQ to achieve good bit rate over the entire transmission bandwidth, and
designs another FIR TEQ to improve the bit rate over a subset of subcarriers. Dualpath
TEQ can be viewed as a special case of a complex valued filter bank structure
that delivers the best bit rate of existing DMT equalizers. However, dual-path
TEQ provides a very good tradeoff between achievable bit rate vs. implementation
complexity on a programmable digital signal processor.Electrical and Computer Engineerin
A robust orthogonal adaptive approach to SISO deconvolution
This paper formulates in a common framework some results from the fields of robust filtering, function approximation with orthogonal basis, and adaptive filtering, and applies them for the design of a general deconvolution processor for SISO systems. The processor is designed to be robust to small parametric uncertainties in the system model, with a partially adaptive orthogonal structure. A simple gradient type of adaptive algorithm is applied to update the coefficients that linearly combine the fixed robust basis functions used to represent the deconvolver. The advantages of the design are inherited from the mentioned fields: low sensitivity to parameter uncertainty in the system model, good numerical and structural behaviour, and the capability of tracking changes in the systems dynamics. The linear equalization of a simple ADSL channel model is presented as an example including comparisons between the optimal nominal, adaptive FIR, and the proposed design.Facultad de IngenieríaComisión de Investigaciones Científicas de la provincia de Buenos Aire
Optimal channel equalization for filterbank transceivers in presence of white noise
Filterbank transceivers are widely employed in data communication networks to cope with inter-symbol-interference (ISI) through the use of redundancies. This dissertation studies the design of the optimal channel equalizer for both time-invariant and time-varying channels, and wide-sense stationary (WSS) and possible non-stationary white noise processes. Channel equalization is investigated via the filterbank transceivers approach. All perfect reconstruction (PR) or zero-forcing (ZF) receiver filterbanks are parameterized in an affine form, which eliminate completely the ISI. The optimal channel equalizer is designed through minimization of the mean-squared-error (MSE) between the detected signals and the transmitted signals. Our main results show that the optimal channel equalizer has the form of state estimators, and is a modified Kalman filter. The results in this dissertation are applicable to discrete wavelet multitone (DWMT) systems, multirate transmultiplexers, orthogonal frequency division multiplexing (OFDM), and direct-sequence/spread-spectrum (DS/SS) based code division multiple access (CDMA) networks. Design algorithms for the optimal channel equalizers are developed for different channel models, and white noise processes, and simulation examples are worked out to illustrate the proposed design algorithms
Development and applications of adaptive IIR and subband filters
Adaptive infinite impulse response (IIR) filter is a challenging research area. Identifiers and Equalizers are among the most essential digital signal processing devices for digital communication systems. In this study, we consider IIR channel both for system identification and channel equalization purposes. We focus on four different approaches: Least Mean Square (LMS), Recursive Least Square (RLS), Genetic Algorithm (GA) and Subband Adaptive Filter (SAF). ). The performance of conventional LMS and RLS based IIR system identification and channel equalization are found with the help of computer simulations. And also the convergence speed and the ability to locate the global optimum solution using a population based algorithm named Genetic Algorithm is given
A New OFDM System for IIR Channels
In this paper, we propose a new OFDM system for an IIR channel with the form
of for two polynomials and . Different from the
conventional OFDM transmission over an FIR channel, a guard interval of an OFDM
symbol is added such that the corresponding part at receiver is the cyclic
prefix (CP) of the received OFDM symbol. The guard interval and CP lengths are
the same and not smaller than the orders of polynomials and . The
OFDM symbol without the guard interval is the same as the conventional OFDM
symbol without the CP. At the receiver, the IIR channel is then converted to
intersymbol interference (ISI) free subchannels, where is the number of
subcarriers of an OFDM symbol
Joint transceiver design for MIMO channel shortening.
Channel shortening equalizers can be employed
to shorten the effective impulse response of a long intersymbol
interference (ISI) channel in order, for example, to decrease the
computational complexity of a maximum-likelihood sequence
estimator (MLSE) or to increase the throughput efficiency of an
orthogonal frequency-division multiplexing (OFDM) transmission
scheme. In this paper, the issue of joint transmitter–receiver filter
design is addressed for shortening multiple-input multiple-output
(MIMO) ISI channels. A frequency-domain approach is adopted
for the transceiver design which is effectively equivalent to an
infinite-length time-domain design. A practical space–frequency
waterfilling algorithm is also provided. It is demonstrated that the
channel shortening equalizer designed according to the time-domain
approach suffers from an error-floor effect. However, the
proposed techniques are shown to overcome this problem and
outperform the time-domain channel shortening filter design. We
also demonstrate that the proposed transceiver design can be considered
as a MIMO broadband beamformer with constraints on
the time-domain multipath length. Hence, a significant diversity
gain could also be achieved by choosing strong eigenmodes of the
MIMO channel. It is also found that the proposed frequency-domain
methods have considerably low computational complexity as
compared with their time-domain counterparts
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