Dissertations / Theses on the topic 'Blind equalization'
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Díguele, Daniel. "Blind equalization /." Online version of thesis, 1994. http://hdl.handle.net/1850/11701.
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Olasz, Elizabeth Barbara. "Blind phase equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp05/nq20763.pdf.
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Abrar, Shafayat. "Blind channel equalization and instantaneous blind source separation." Thesis, University of Liverpool, 2010. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.540044.
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Yang, Jian. "Multimodulus algorithms for blind equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq25191.pdf.
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Kutlu, Mehmet. "Kalman filtering approach to blind equalization." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1993. http://handle.dtic.mil/100.2/ADA276320.
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Lim, Sze Chie (Felicia). "Robust multichannel equalization for blind speech dereverberation." Thesis, Imperial College London, 2016. http://hdl.handle.net/10044/1/39566.
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Acoustic reverberation arises from the reflection of sound waves within an enclosed space. It is generally desirable in music reproduction but can be detrimental to speech-related applications. For the human listener, while the early reflections help to improve speech intelligibility, the late reflections have been shown to impair perceived speech quality. For speech processing technologies such as automatic speech recognizers, reverberation reduces accuracy and performance. Dereverberation is therefore an important research topic with interest driven by increasing availability of communication devices and consumer demand. One approach to dereverberation computes a set of equalizing filters that are used to perform the dereverberation processing, given multichannel inputs and estimates of the acoustic impulse responses (AIRs) between the source signal and microphones. However, estimation errors are inevitable in practice and therefore robust channel equalizers are required. This thesis aims to develop such robust algorithms in a manner that is desirable specifically for speech dereverberation. The framework of channel shortening is used, having been previously shown to give promising results. Subband approaches are also investigated to reduce the computational complexity and achieve finer control of dereverberation in separate frequency bands. A second approach to dereverberation steers the look direction of beamformers towards the source. Reverberant sounds from other directions are treated as noise and accordingly suppressed. The motivation behind beamformer design and channel equalization is similar and in this work, a unified framework termed MINTFormer is proposed. The aim is to combine the robustness of beamformers with the potentially perfect dereverberation ability that can be achieved by channel equalization approaches.
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Jain, Vijay. "A new sufficient-order blind equalization scheme." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape8/PQDD_0006/MQ43652.pdf.
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Lambotharan, Sangarapillai. "Algorithms and structures for adaptive blind equalization." Thesis, Imperial College London, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.268038.
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Adnan, Rubyet. "Blind Equalization for Tomlinson-Harashima Precoded Systems." Thesis, University of Canterbury. Electrical and Computer Engineering, 2007. http://hdl.handle.net/10092/1130.
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At a communications receiver the observed signal is a corrupted version of the transmitted signal. This distortion in the received signal is due to the physical characteristics of the channel, including multipath propagation, the non-idealities of copper wires and impulse noise. Equalization is a process to combat these distortions in order to recover the original transmitted signal. Roughly stated, the equalizer tries to implement the inverse transfer function of the channel while taking into account the channel noise. The equalizer parameters can be tuned to this inverse transfer function using an adaptive algorithm. In many cases, the algorithm uses a training sequence to drive the equalizer parameters to the optimum solution. But, for time-varying channels or multiuser channels the use of a training sequence is inefficient in terms of bandwidth, as bandwidth is wasted due to the periodic re-transmission of the training sequence. A blind equalization algorithm is a practical method to eliminate this training sequence. An equalizer adapted using a blind algorithm is a key component of a bandwidth efficient receiver for broadcast and point-to-multipoint communications. The initial convergence performance of a blind adaptive equalizer depends on the higher-order statistics of the transmitted signal. In modern digital systems, Tomlinson-Harashima precoding (THP) is often used for signal shaping and to mitigate the error propagation problem of a decision feedback equalizer (DFE). The concept of THP comes from pre-equalization. In fact, it is a nonlinear form of pre-equalization, which bounds the higher-order statistics of the transmitted signal. But, THP and blind equalization are often viewed as incompatible equalization techniques. In this research, we give multiple scenarios where blind equalization of a THP-encoded signal might arise. With this motivation we set out to answer the question, can a blind equalizer successfully acquire a THP-encoded signal? We investigate the combination of a Tomlinson-Harashima precoder on the transmitter side and a blind equalizer on the receiver side. By bounding the kurtosis of the THP-encoded signal, we show that THP actually aids the initial convergence of blind equalization. We find that, as the symbol constellation size increases, the THP-encoded signal kurtosis approaches that of a uniform distribution, not a Gaussian. We investigate the compatibility of blind equalization with THP-encoded signals for both SISO and MIMO systems. In a SISO system, conventional blind algorithms can be used to counter the distortions introduced in the received signal. However, in a MIMO system with multiple users, the other users act as interferers on the desired user's signal. Hence, modified blind algorithms need to be applied to mitigate these interferers. For both SISO and MIMO systems, we show that the THP encoder ensures that the signal distribution approaches a non-Gaussian distribution. Using Monte Carlo simulations, we study the effects of Tomlinson-Harashima precoding on the performance of Bussgang-type blind algorithms and verify our theoretical analysis. The major contributions of this thesis are: • A demonstration that a blind equalizer can successfully acquire a THP-encoded signal for both SISO and MIMO systems. We show that THP actually aids blind equalization, as it ensures that the transmitted signal is non-Gaussian. • An analytical quantification of the effects of THP on the transmitted signal statistics. We derive a novel bound on the kurtosis of the THP-encoded signal. • An extension of the results from a single-user SISO scenario to multiple users and a MIMO scenario. We demonstrate that our bound and simulated results hold for these more general cases. Through our work, we have opened the way for a novel application of training sequence-less equalization: to acquire and equalize THP-encoded signals. Using our proposed system, periodic training sequences for a broadcast or point-to-multipoint system can be avoided, improving the bandwidth efficiency of the transceiver. Future modem designs with THP encoding can make use of our advances for bandwidth efficient communication systems.
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Xanthopoulos, Spyridon. "A novel initialization scheme for blind equalization algorithms." Thesis, Imperial College London, 2002. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.418896.
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Ramkumar, Barathram. "Automatic Modulation Classication and Blind Equalization for Cognitive Radios." Diss., Virginia Tech, 2011. http://hdl.handle.net/10919/28666.
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Cognitive Radio (CR) is an emerging wireless communications technology that addresses the inefficiency of current radio spectrum usage. CR also supports the evolution of existing wireless applications and the development of new civilian and military applications. In military and public safety applications, there is no information available about the signal present in a frequency band and hence there is a need for a CR receiver to identify the modulation format employed in the signal.
The automatic modulation classifier (AMC) is an important signal processing component that helps the CR in identifying the modulation format employed in the detected signal.
AMC algorithms developed so far can classify only signals from a single user present in a frequency band. In a typical CR scenario, there is a possibility that more than one user is present in a frequency band and hence it is necessary to develop an AMC that can classify signals from multiple users simultaneously. One of the main objectives of this dissertation is to develop robust multiuser AMC's for CR. It will be shown later that multiple antennas are required at the receiver for classifying multiple signals. The use of multiple antennas at the transmitter and receiver is known as a Multi Input Multi Output (MIMO) communication system. By using multiple antennas at the receiver, apart from classifying signals from multiple users, the CR can harness the advantages offered by classical MIMO communication techniques like higher data rate, reliability, and an extended coverage area. While MIMO CR will provide numerous benefits, there are some significant challenges in applying conventional MIMO theory to CR. In this dissertation, open problems in applying classical MIMO techniques to a CR scenario are addressed.
A blind equalizer is another important signal processing component that a CR must possess since there are no training or pilot signals available in many applications. In a typical wireless communication environment the transmitted signals are subjected to noise and multipath fading. Multipath fading not only affects the performance of symbol detection by causing inter symbol interference (ISI) but also affects the performance of the AMC. The equalizer is a signal processing component that removes ISI from the received signal, thus improving the symbol detection performance. In a conventional wireless communication system, training or pilot sequences are usually available for designing the equalizer. When a training sequence is available, equalizer parameters are adapted by minimizing the well known cost function called mean square error (MSE). When a training sequence is not available, blind equalization algorithms adapt the parameters of the blind equalizer by minimizing cost functions that exploit the higher order statistics of the received signal. These cost functions are non convex and hence the blind equalizer has the potential to converge to a local minimum. Convergence to a local minimum not only affects symbol detection performance but also affects the performance of the AMC. Robust blind equalizers can be designed if the performance of the AMC is also considered while adapting equalizer parameters. In this dissertation we also develop Single Input Single Output (SISO) and MIMO blind equalizers where the performance of the AMC is also considered while adapting the equalizer parameters.Ph. D.
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Fung, Carrson Chee-Ho. "Eigensystem based techniques for blind channel estimation and equalization /." View abstract or full-text, 2005. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202005%20FUNG.
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Zhang, Wen Pian Paul. "Transmitter-induced cyclostationarity for blind channel identification and equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0019/MQ49753.pdf.
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Redfern, Arthur John. "Data aided and blind equalization of nonlinear communication channels." Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/13446.
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Au, Kelvin Kar-Kin. "Semi-blind spatial-temporal equalization for short burst wireless communications." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0021/MQ53423.pdf.
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Stranges, Anthony G. "Blind equalization and fading channel signal recovery of OFDM modulation." Thesis, Monterey, California. Naval Postgraduate School, 2011. http://hdl.handle.net/10945/5740.
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Approved for public release; distribution is unlimited.Algorithms for blind equalization and data recovery of orthogonal frequency-division multiplexed (OFDM) signals transmitted through fading channels are implemented and simulated in this thesis. The channel is estimated without knowledge of the transmitted sequence (i.e., blindly) using a least mean squares (LMS) adaptive filter and filter bank precoders. This method was used to estimate channel characteristics using both binary and quadrature phase-shift keying signals. Additionally, the method was analyzed for robustness with a poor initial estimate of channel characteristics, with the addition of white Gaussian noise to the signal, and with non-stationary channel conditions. In addition, it is shown that the proposed method is particularly suited in situations with deep fading channels, where some of the subcarriers have a very low SNR. Simulations for both aspects of this thesis were conducted using MATLAB, and the results are presented.
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Ye, Zhong, Edgar H. Satorius, Thomas C. Jedrey, and Kip Temple. "Enhancement of Advanced Range Telemetry (ARTM) Channels via Blind Equalization." International Foundation for Telemetering, 2001. http://hdl.handle.net/10150/607696.
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International Telemetering Conference Proceedings / October 22-25, 2001 / Riviera Hotel and Convention Center, Las Vegas, NevadaThe Joint Services Advanced Range Telemetry (ARTM) Program at Edwards Air Force Base has been evaluating FQPSK-B for possible upgrades to the existing telemetry equipment. It has been found in the wideband channel sounding experiments sponsored by ARTM that the in-flight fading channel can be modeled as a 3-ray multipath channel[1]. Delay spread for a typical in-flight channel is in the order of 300 nanoseconds. Furthermore, the pre-flight channel is characterized by much more severe multipath, in which the delay spread is in the order of microseconds covering one or more symbols when the FQPSK-B transceiver operates at a rate of millions of symbols per second. This adverse channel condition inevitably causes tremendous distortion in the received signals due to severe inter-symbol interference (ISI) from the multipath. This paper provides an assessment of the potential ability of blind equalization to reduce the FQPSK-B system susceptibility to degradation caused by dynamic frequency selective fading in the aeronautical telemetry environment. In particular, a blind equalizer applique that can be inserted prior to the demodulator without knowledge of the received signal such as carrier frequency, symbol timing and sequence, etc, is proposed. Since it is desired that the equalizer applique operate independently of the carrier frequency and given that the modulation of interest is constant envelope (PCM-FM or FQPSKB), we have selected the constant modulus algorithm (CMA)[2] cost function for implementation. Extensive tests on both simulated and recorded FQPSK-B data transmitted over different ARTM channels have been conducted and the blind equalizer structure has shown substantial improvements, even on the difficult ARTM pre-flight channels. The CMA adapts the equalizer coefficients to minimize the deviation of the output envelope from an arbitrary constant level. This paper depicts the pre-flight and in-flight channel conditions using time and spectral domain measurement. It quantifies the benefit of the blind CMA tapped delay line equalizer. Due to the extensive signal processing requirements associated with the very high sampling rate (100 MHz) of the FQPSK-B system, hardware implementation complexity is very high. Complexity reduction issues regarding the implementation of the CMA using Field Programmable Gate Array (FPGA) will also be presented.
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Kennedy, Rodney Andrew, and rodney kennedy@anu edu au. "Operational Aspects of Decision Feedback Equalizers." The Australian National University. Research School of Physical Sciences and Engineering, 1989. http://thesis.anu.edu.au./public/adt-ANU20050418.151329.
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The central theme is the study of error propagation effects in decision feedback equalizers (DFEs). The thesis contains: a stochastic analysis of error propagation in a tuned DFE; an analysis of the effects of error propagation in a blindly adapted DFE; a deterministic analysis of error propagation through input-output stability ideas; and testing procedures for establishing correct tap convergence in blind adaptation. To a lesser extent, the decision directed equalizer (DDE) is also treated.¶ Characterizing error propagation using finite state Markov process (FSMP) techniques is first considered. We classify how the channel and DFE parameters affect the FSMP model and establish tight bounds on the error probability and mean error recovery time of a tuned DFE. These bounds are shown to be too conservative for practical use and highlight the need for imposing stronger hypotheses on the class of channels for which a DFE may be effectively used.¶ In blind DFE adaptation we show the effect of decision errors is to distort the adaptation relative to the use of a training sequence. The mean square error surface in a LMS type setting is shown to be a concatenation of quadratic functions exposing the possibility of false tap convergence to undesirable DFE parameter settings. Averaging analysis and simulation are used to verify this behaviour on some examples.¶ Error propagation in a tuned DFE is also examined in a deterministic setting. A finite error recovery time problem is set up as an input-output stability problem. Passivity theory is invoked to prove that a DFE can be effectively used on a channel satisfying a simple frequency domain condition. These results give performance bounds which relate well with practice.¶ Testing for false tap convergence in blind adaptation concludes our study. Simple statistic output tests are shown to be capable of discerning correct operation of a DDE. Similar tests are conjectured for the DFE, supported by proofs for the low dimensional cases.
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Karakutuk, Serkan. "Blind And Semi-blind Channel Order Estimation In Simo Systems." Phd thesis, METU, 2009. http://etd.lib.metu.edu.tr/upload/12611107/index.pdf.
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Channel order estimation is an important problem in many fields including signal processing,
communications, acoustics, and more. In this thesis, blind channel order estimation problem
is considered for single-input, multi-output (SIMO) FIR systems. The problem is to estimate
the effective channel order for the SIMO system given only the output samples corrupted by
noise. Two new methods for channel order estimation are presented. These methods have
several useful features compared to the currently known techniques. They are guaranteed to
find the true channel order for noise free case and they perform significantly better for noisy
observations. These algorithms show a consistent performance when the number of observations,
channels and channel order are changed. The proposed algorithms are integrated with
the least squares smoothing (LSS) algorithm for blind identification of the channel coefficients. LSS algorithm is selected since it is a deterministic algorithm and has some additional
features suitable for order estimation. The proposed algorithms are compared with a variety
of dierent algorithms including linear prediction (LP) based methods. LP approaches are
known to be robust to channel order overestimation. In this thesis, it is shown that significant
gain can be obtained compared to LP based approaches when the proposed techniques are
used. The proposed algorithms are also compared with the oversampled single-input, single-output (SISO) system with a generic decision feedback equalizer, and better mean-square
error performance is observed for the blind setting.
Channel order estimation problem is also investigated for semi-blind systems where a pilot
signal is used which is known at the receiver. In this case, two new methods are proposed
which exploit the pilot signal in dierent ways. When both unknown and pilot symbols are
used, a better estimation performance can be achieved compared to the proposed blind methods.
The semi-blind approach is especially effective in terms of bit error rate (BER) evaluation
thanks to the use of pilot symbols in better estimation of channel coecients. This approach
is also more robust to ill-conditioned channels. The constraints for these approaches, such
as synchronization, and the decrease in throughput still make the blind approaches a good
alternative for channel order estimation. True and effective channel order estimation topics
are discussed in detail and several simulations are done in order to show the significant performance
gain achieved by the proposed methods.
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Fathallah, Habib. "Narrowband interference suppression in spread spectrum CDMA communications via blind equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/mq25573.pdf.
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Tse, Eloise. "Blind equalization with differential detection for channels with ISI and fading." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk2/ftp01/MQ28855.pdf.
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Tian, Yunhao. "Subspace method for blind equalization of multiple time-varying FIR channels." Thesis, McGill University, 2012. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=107827.
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Wireless communications is the fastest growing segment of communication technologies. In a wireless communication system, the inter-symbol interference (ISI) is a linear distortion which causes decision errors at the receiver. The equalizer is required to remove the ISI. In the past decade, the blind channel equalization has been a popular research topic in the area of wireless communication. A particular class of blind equalization approaches is based on the second order statistics (SOS) of the received signals. Within this framework, subspace methods exploit the orthogonality between the signal and noise subspaces in order to identify the channel characteristics so that the equalizer can be constructed.This thesis investigates a new equalization algorithm for the time-varying (TV) channel under the single-input multiple-output (SIMO) framework. The TV channel is decomposed using arbitrary basis functions associated with time variable properties of the channels, and with expansion coefficients associated with multi-path delays. An equivalent time-invariant (TI) multiple-input multiple-output (MIMO) system is built for the TV SIMO system. The equivalent TI MIMO system is assumed to match the necessary and sufficient conditions of the SOS identification framework. The blind subspace method is exploited to identify the expansion coefficients when considered as channel characteristics of the MIMO system. The associated ambiguity matrix is identified by using the least square (LS) method. The zero forcing equalizer is realized based on the result of the subspace channel equalization and the ambiguity matrix. The simulation results indicate that the proposed equalizer can effectively recover the source signal in TV SIMO channel applications.La communication sans fil est le segment de croissance le plus dynamique parmi les techniques de la communication. Dans un système de communication sans fil, l'interférence inter-symboles (ISI) est une distorsion linéaire qui provoque des erreurs de décisions au niveau du récepteur. L'égaliseur est nécessaire pour éliminer l'ISI. Récemment, l'égalisation aveugle du canal est devenue un sujet de recherche populaire dans les domaines de la communication sans fil. Un des jalons de la technologie aveugle est fondé sur le cadre des statistiques du second ordre (SOS) du signal reçu. Tout particulièrement, la méthode du sous-espace exploite l'orthogonalité entre le sous-espace signal et le sous-espace bruit afin d'identifier les caractéristiques du canal de telle sorte que l'égaliseur puisse être construit. Dans cette thèse, j'ai proposé un algorithme de péréquation pour le canal à variation temporelle (TV) des systèmes à entrée unique et sorties multiples (SIMO). Le canal TV est décomposé en fonctions arbitraires associées aux propriétés de TV du cacal, et avec les coefficients d'expansion associés à chacun des retards multi-trajet. Un système équivalent invariant dans le temps (TI), à entrée multiples et sorties multiples (MIMO) est conçu pour le TV SIMO. Le systèmeéquivalent TI MIMO est supposé correspondre aux conditions nécessaires et suffisantes dans le cadre de la théorie SOS. La méthode sous-espace aveugle est exploitée pour identifier les coefficients d'expansion quand ils sont considérés comme caractéristiques du canal du système MIMO. La matrice d'ambiguïté est déterminée par la méthode des moindres carrés (LS). La remise à zéro forcée de l'égaliseur est réalisée sur la base des résultats de l'égalisation des canaux de sous-espace et de la matrice d'ambiguïté. Des expériences de simulations numériques sont utilisées afin de démontrer le potential d'application de la nouvelle méthode.
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Blanc, Grégory, Alexandre Skrzypczak, and Jean-Guy Pierozak. "BLIND EQUALIZATION WITH LDPC CODE: TO QUASIERROR FREE TRANSMISSIONS IN TELEMETRY." International Foundation for Telemetering, 2016. http://hdl.handle.net/10150/624248.
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In a telemetry system, it has been frequently proved that multipath channels and transmission
noise are the most critical sources of distortion. While equalization allows a strong limitation of
the multipath effects, the noise impact can be efficiently reduced if forward error correction is
used. This paper proves that the combination of blind equalization and a powerful FEC like
LDPC strongly improves bit error rates for the SOQPSK modulation. We also prove that a LDPC
code is able to fully correct the residual errors that may persist at the equalizer output. In other
terms, the combination of equalization and LDPC code enables quasi-error free transmissions in
various channel scenarios that represent the various phases of a telemetry mission.
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Nguyen, Hoang. "The expectation-maximization Viterbi algorithm for blind channel identification and equalization /." For electronic version search Digital dissertations database. Restricted to UC campuses. Access is free to UC campus dissertations, 2003. http://uclibs.org/PID/11984.
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Senol, Sinan. "Performance Comparison Of Adaptive Decision Feedback Equalizer And Blind Decision Feedback Equalizer." Master's thesis, METU, 2004. http://etd.lib.metu.edu/upload/1023746/index.pdf.
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The Decision Feedback Equalizer (DFE) is a known method of channel equalization which has performance superiority over linear equalizer. The best performance of DFE is obtained, commonly, with training period which is used for initial acquisiton of channel or recovering changes in the channel. The training period requires a training sequence which reduces the bit transmission rate or is not possible to send in most of the situations. So, it is desirable to skip the training period. The Unsupervised (Blind) DFE (UDFE) is such a DFE scheme which has no training period. The UDFE has two modes of operation. In one mode, the UDFE uses Constant Modulus Algorithm (CMA) to perform channel acquisition, blindly. The other mode is the same as classical decision-directed DFE. This thesis compares the performances of the classical trained DFE method and the UDFE. The performance comparison is done in some channel environments with the problem of timing error present in the received data bearing signal. The computer aided simulations are done for two stationary channels, a time-varying channel and a frequency selective Rayleigh fading channel to test the performance of the relevant equalizers. The test results are evaluted according to mean square error (MSE), bit-error rate (BER), residual intersymbol interference (RISI) performances and equalizer output diagrams. The test results show that the UDFE has an equal or, sometimes, better performance compared to the trained DFE methods. The two modes of UDFE enable it to solve the absence of training sequence.
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Roy, Pulakesh. "Fractionally Spaced Blind Equalizer Performance Improvement." Thesis, Virginia Tech, 2000. http://hdl.handle.net/10919/31048.
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Blind equalization schemes are used to cancel the effects of a channel on the received signal when the transmission of a training sequence in a predefined time slot is not possible. In the absence of a training sequence, blind equalization schemes can also increase the throughput of the overall system. A general problem with blind adaptation techniques is that they have poor convergence properties compared to the traditional techniques using training sequences. Having a multi-modal cost surface, blind adaptation techniques may force the equalizer to converge to a false minimum, depending on the initialization. The most commonly used blind adaptation algorithm is the Constant Modulus Algorithm (CMA). It is shown by simulation that a logarithmic error equation can make CMA converge to a global minimum, if a differential encoding scheme is used. The performance of CMA with different error equations is also investigated for different channel conditions.
For a time varying channel, the performance of an equalizer not only depends on the convergence behavior but also on the tracking property, which indicates the ability of an equalizer to track changes in the channel. The tracking property of a blind equalizer with CMA has been investigated under different channel conditions. It is also shown that the tracking property of a blind equalizer can be improved by using a recursive linear predictor at the output of the equalizer to predict the amplitude of the equalizer output. The predicted value of the amplitude is then used to adjust the instantaneous gain of the overall system.
A recursive linear predictor is designed to predict a colored signal without having a priori knowledge about the correlation function of the input sequence. The performance of the designed predictor is also investigated by predicting the envelope of a flat fading channel under constant mobile velocity and constant acceleration conditions.Master of Science
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Jiang, Yufei. "Semi-blind CFO estimation and ICA based equalization for wireless communication systems." Thesis, University of Liverpool, 2014. http://livrepository.liverpool.ac.uk/19095/.
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In this thesis, a number of semi-blind structures are proposed for Orthogonal Frequency Division Multiplexing (OFDM) based wireless communication systems, with Carrier Frequency Offset (CFO) estimation and Independent Component Analysis (ICA) based equalization. In the first contribution, a semi-blind non-redundant single-user Multiple-Input Multiple-Output (MIMO) OFDM system is proposed, with a precoding aided CFO estimation approach and an ICA based equalization structure. A number of reference data sequences are carefully designed and selected from a pool of orthogonal sequences, killing two birds with one stone. On the one hand, the precoding based CFO estimation is performed by minimizing the sum cross-correlations between the CFO compensated signals and the rest of the orthogonal sequences in the pool. On the other hand, the same reference data sequences enable the elimination of permutation and quadrant ambiguities in the ICA equalized signals. Simulation results show that the proposed semi-blind MIMO OFDM system can achieve a Bit Error Rate (BER) performance close to the ideal case with perfect Channel State Information (CSI) and no CFO. In the second contribution, a low-complexity semi-blind structure, with a multi-CFO estimation method and an ICA based equalization scheme, is proposed for multiuser Coordinated Multi-Point (CoMP) OFDM systems. A short pilot is carefully designed offline for each user and has a two-fold advantage. On the one hand, using the pilot structure, a complex multi-dimensional search for multiple CFOs is divided into a number of low-complexity mono-dimensional searches. On the other hand, the cross-correlation between the transmitted and received pilots is explored to allow the simultaneous elimination of permutation and quadrant ambiguities in the ICA equalized signals. Simulation results show that the proposed semi-blind CoMP OFDM system can provide a BER performance close to the ideal case with perfect CSI and no CFO. In the third contribution, a semi-blind structure is proposed for Carrier Aggregation (CA) based CoMP Orthogonal Frequency Division Multiple Access (OFDMA) systems, with an ICA based joint Inter-Carrier Interference (ICI) mitigation and equalization scheme. The CFO-induced ICI is mitigated implicitly via ICA based equalization, without introducing feedback overhead for CFO correction. The permutation and quadrant ambiguities in the ICA equalized signals can be eliminated by a small number of pilots. Simulation results show that with a low training overhead, the proposed semi-blind equalization scheme can provide a BER performance close to the ideal case with perfect CSI and no CFO.
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Lee, Zhi Hou. "Improved multiple input multiple output blind equalization algorithms for medical implant communication." Thesis, University of Nottingham, 2015. http://eprints.nottingham.ac.uk/28726/.
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Medical implant sensor that is used to monitor the human physiology signals is helpful to improve the quality of life and prevent severe result from the chronic diseases. In order to achieve this, the wireless implant communication link that delivers the monitored signal to a multiple antennas external device is an essential portion. However, the existing conventional narrow band Medical Implant Communications System (MICS) has low data rate because of the bandlimited channel is allocated. To improve the data rate in the radio frequency communication, ultra-wide band technology has been proposed. However, the ultra-wide band technology is relatively new and requires living human to be the test subject in order to validate the technology performance. In this condition, the test on the new technology can rise ethical challenge. As a solution, we improve the data rate in the conventional narrow band MICS. The improvement of data rate on the narrow band implies the information bandwidth is larger than the allocated channel bandwidth, and therefore the high frequency components of the information can loss. In this case, the signal suffers the intersymbol-interference (ISI). Instead of that, the multiple antennas external device can receive the signal from other transmitting implant sensor which has the same operating frequency. As a result, the signal is further hampered by co-channel interference (CCI). To recover the signal from the ISI and CCI, multiple-input multiple output (MIMO) blind equalization that has source separation ability can be exploited. Cross-Correlation Constant Modulus Algorithm (CC-CMA) is the conventional MIMO blind equalization algorithm that can suppress ISI and CCI and able to perform source separation. However, CC-CMA has only been analyzed and simulated in the modulation of Phase Shift Keying (PSK). The performance of CC-CMA in multi-modulus modulation scheme such as 4-Pulse-amplitude modulation (PAM) and 16-Quadrature amplitude modulation (QAM), which has higher data rate than PSK, has not been analyzed. Therefore, our work is to analysis and optimize CC-CMA on the multi-modulus modulation scheme. From our analysis, we found that the cost function of CC-CMA is biased cost function. Instead of that, from our simulation, CC-CMA introduces an unexpected shrinking effect whereby the amplitudes of the equalizer outputs have been reduced, especially in multi-modulus modulation scheme. This shrinking effect is not severe in PSK because the decision of a PSK symbol is based on phase, but not amplitude. Unfortunately, this is severe in multi-modulus modulation scheme. To overcome this shrinking effect in multi-modulus modulation scheme, we propose Cross-Independent Constant Modulus Algorithm (CI-CMA). Based on the convergence analysis, we identify the new optimum dispersion value and mixing parameter in CI-CMA. From the simulation results, we confirm that CI-CMA is able to perform equalization and source separation in the multi-modulus modulation scheme. In order to improve the steady state performance of CI-CMA, we perform the steady state mean square error (MSE) analysis of CI-CMA using the energy preservation theorem that was developed by Mai and Sayed in 2001, and our result is more accurate than the previous work. From our analysis, only the reduction in adaptation step size can reduce the steady state MSE, but it is well known that the MSE is indeed a tradeoff with the speed of convergence. Therefore without sacrificing convergence speed, our last effort is to propose hybrid algorithms. The hybrid algorithms are done by combining a new adaptive constant modulus algorithm (ACMA), a decision directed algorithm and a cross-correlation function. From the simulation results, we found that the hybrid algorithms can show low steady state error and thereby improve the reliability of the communication link. The main achievement of this thesis is the discovery of new dispersion value through the convergence analysis.
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Gao, Wei, Shih-Ho Wang, and Kamilo Feher. "BLIND EQUALIZATION FOR FQPSK AND FQAM SYSTEMS IN MULTIPATH FREQUENCY SELECTIVE FADING CHANNELS." International Foundation for Telemetering, 1999. http://hdl.handle.net/10150/607298.
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International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, NevadaBlind adaptive equalization with application for Non-Linearly Amplified (NLA) quadrature amplitude modulation (QAM) systems in multipath selective fading channels is presented. With an offset sampling strategy in the receiver, the proposed blind equalization using Constant Modulus Algorithm (CMA) exhibits a fast convergent speed for a family of quadrature modulated systems in NLA and multipath fading channels. Feher’s patented Quadrature Phase Shift Keying (FQPSK) and Feher’s Quadrature Amplitude Modulation (FQAM) which correspond respectively to 4-state and 16-state QAM are used due to their higher Radio Frequency (RF) power and spectral efficiency in NLA channel. It has been shown that blind adaptive equalization can significantly open the eye signals in multipath frequency selective fading channels.
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Zhu, Weihua. "Blind Equalization and Multi-User Detection in Asynchronous Space-Time Coded CDMA System." Cincinnati, Ohio : University of Cincinnati, 2004. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=ucin1092948594.
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KoneDossongui, Serge, Oladotun Opasina, Henry Umuolo, Habtamu Betelle, Solomon Thang, and Robin Shrestha. "Evaluation of CMA+AMA Equalization for SOQPSK Modulation in Aeronautical Telemetry." International Foundation for Telemetering, 2013. http://hdl.handle.net/10150/579587.
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ITC/USA 2013 Conference Proceedings / The Forty-Ninth Annual International Telemetering Conference and Technical Exhibition / October 21-24, 2013 / Bally's Hotel & Convention Center, Las Vegas, NVMultipath interference continues to be the dominant cause of telemetry link outages in low-elevation angle reception scenarios. The most reliable and universally applicable solution to this problem is in the form of equalization. Previous work in this area has considered the Constant modulus algorithm (CMA) equalizer operating in a blind adaptive mode. To the extent that knowledge of the multipath channel improves the performance of CMA and related equalizers and permits the use of other equalization techniques, data aided equalizers are of interest. Channel knowledge is obtained by comparing the received samples with the samples corresponding to a known bit pattern (called a pilot block) periodically inserted in the telemetry data stream. The main objective of this research is to evaluate the performance of a modified CMA equalization algorithm, which has the property of automatically resolving the phase of the QPSK modulated symbol, and to determine its suitability for use with SOQPSK-TG by taking into account the capability of exploiting the presence of a periodically inserted pilot block. As an initial effort in that direction, this paper provides simulation results of the error performance of the blind linear combination of CMA and alphabet matched algorithm (AMA) equalizer as compared to that of pilot assisted equalization with SOQPSK modulation over aeronautical channel.
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Minardi, Michael Joseph. "Blind adaptive signal processing with applications to channel equalization and WDM fiber-optic receivers." Diss., Georgia Institute of Technology, 1996. http://hdl.handle.net/1853/16356.
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Skrzypczak, Alexandre, Grégory Blanc, Bournault Tangi Le, and Jean-Guy Pierozak. "Limitation of the 2-Antennas Problem for Aircraft Telemetry by Using a Blind Equalizer." International Foundation for Telemetering, 2015. http://hdl.handle.net/10150/596457.
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ITC/USA 2015 Conference Proceedings / The Fifty-First Annual International Telemetering Conference and Technical Exhibition / October 26-29, 2015 / Bally's Hotel & Convention Center, Las Vegas, NVThe emission of the telemetry signal is required over minimum two different antennas to keep the telemetry link available during a maneuver of a flying object. If nothing is made at the transmitter side, the telemetry link can be fully lost as both signals may have an opposite phase. We here propose a simple solution based on delay diversity to solve this problem. The basic idea is to introduce a delay between both emitted signals to guarantee a non-destructive signal recombination. We then exploit the ability of the blind equalizer developed by ZDS for the PCM/FM modulation to correctly equalize this signal and to recover the initial data. This solution does not require any modification of the on-board and floor set-ups except the introduction of a delay line between both transmitting antennas. It also does not need any pilot sequence and is natively robust to multipath perturbations.
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Ahmed, Rana [Verfasser], and Joachim [Akademischer Betreuer] Speidel. "Equalization and blind signal combining algorithms for mobile television broadcast reception / Rana Ahmed ; Betreuer: Joachim Speidel." Stuttgart : Universitätsbibliothek der Universität Stuttgart, 2016. http://d-nb.info/1118507460/34.
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Terziev, George, and Kamilo Feher. "ADAPTIVE FAST BLIND FEHER EQUALIZERS (FE) FOR FQPSK." International Foundation for Telemetering, 1999. http://hdl.handle.net/10150/607297.
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International Telemetering Conference Proceedings / October 25-28, 1999 / Riviera Hotel and Convention Center, Las Vegas, NevadaThe performance of novel experimental blind equalizers suitable for a large class of applications including telemetry systems and other wireless applications is described. Experimental hardware research of these adaptive patent pending Feher Equalizers (FE) confirms computer simulated data [1]. A two-ray RF selective faded telemetry channel has been simulated. A dynamically changing channel environment with a selective fade rate in the 1Hz to 50Hz range has been constructed by laboratory hardware. The Test and Evaluation (T&E) setup had RF frequency selective dynamic notch depth variations in the Power Spectral Density (PSD) within the band of the signal of up to 15dB. As an illustrative example of the adaptive equalizer capability we used a 1Mb/s rate Feher patented FQPSK [1] Commercially Of The Shelf (COTS) product. Both hardware experimental results as well as simulation indicate substantial performance improvement with the utilization of the FE. It is demonstrated that the FE improves for a large class of frequency selective faded systems the Bit Error Rate(BER) from 10^-2 to 10^-6. Similar performance improvements are presented for the Block Error Rate (BLER).
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Skrzypczak, Alexandre, Grégory Blanc, and Bournault Tangi Le. "Combining a Reed-Solomon Block Code with a Blind Equalizer: Synchronization and Bit Error Rate Performance." International Foundation for Telemetering, 2015. http://hdl.handle.net/10150/596403.
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ITC/USA 2015 Conference Proceedings / The Fifty-First Annual International Telemetering Conference and Technical Exhibition / October 26-29, 2015 / Bally's Hotel & Convention Center, Las Vegas, NVThe performance of telemetry systems may be strongly affected by diverse sources of perturbations. Among them, multipath channels and transmission noise are the most critical. While the effects due to the multipath channels can be attenuated thanks to equalization, the effects of the noise are limited if forward error correction is used. This paper first proves that the combination of blind equalization and forward error correction can strongly improve bit error rates. The other objective of the paper is to show that reasonably powerful codes like Reed-Solomon codes are sufficient to enable quasi-error free transmissions in a large majority of propagation channel scenarios.
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Silva, Magno Teófilo Madeira da. "Um estudo sobre técnicas de equalização autodidata." Universidade de São Paulo, 2005. http://www.teses.usp.br/teses/disponiveis/3/3142/tde-01042005-214250/.
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Neste trabalho, investigam-se técnicas autodidatas baseadas em estatísticas de ordem superior, aplicadas à equalização de canais de comunicação. Inicialmente, obtém-se um intervalo do passo de adaptação que assegura a convergência do algoritmo do Módulo Constante com o gradiente exato. Algoritmos como o CMA (Constant Modulus Algorithm) e o SWA (Shalvi-Weinstein Algorithm) são revisitados e suas capacidades de tracking analisadas, utilizando-se uma relação de conservação de energia. Além disso, é proposto um algoritmo autodidata denominado AC-CMA (Accelerated Constant Modulus Algorithm) que utiliza a segunda derivada (aceleração) da estimativa dos coeficientes. Esse algoritmo pode apresentar um compromisso mais favorável entre complexidade computacional e velocidade de convergência que o CMA e o SWA. Esses resultados são estendidos para o caso multiusuário. Através de simulações, os algoritmos são comparados e as análises de convergência e tracking validadas. Considerando o DFE (Decision Feedback Equalizer) no caso monousuário com o critério do módulo constante, é proposto um algoritmo concorrente que evita soluções degeneradas e apresenta um desempenho melhor do que os existentes na literatura. Com o intuito de evitar propagação de erros, é proposta uma estrutura híbrida que utiliza uma rede neural recorrente na malha de realimentação. Resultados de simulações indicam que seu uso pode ser vantajoso para canais lineares e não-lineares.The equalization of communication channels is addressed by using blind techniques based on higher order statistics. A step-size interval is obtained to ensure the convergence of Steepest-Descent Constant Modulus Algorithm. The Shalvi-Weinstein Algorithm (SWA) and Constant Modulus Algorithm (CMA) are revisited and their tracking capabilities are analyzed by using an energy conservation relation. Moreover, a novel blind algorithm named Accelerated Constant Modulus Algorithm (AC-CMA) is proposed. It adjusts the second derivative (acceleration) of the coefficient estimates and presents a more favorable compromise between computational complexity and convergence rate than CMA or SWA. These results are extended to the MIMO (Multiple-Input Multiple-Output) case. By means of simulations, the algorithms are compared and the convergence and tracking analysis are validated. The Decision Feedback Equalizer (DFE) is considered in the SISO (Single-Input Single-Output) case with the Constant Modulus criterion and a concurrent algorithm is proposed. It avoids degenerated solutions and shows better behavior than the others presented in the literature. In order to avoid error propagation, a hybrid DFE is also proposed. It includes a recurrent neural network in the feedback filter and may be advantageously used to equalize linear and nonlinear channels.
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Tang, Sze Ho. "Self-correcting multi-channel Bussgang blind deconvolution using expectation maximization (EM) algorithm and feedback." Diss., Georgia Institute of Technology, 2009. http://hdl.handle.net/1853/33975.
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A Bussgang based blind deconvolution algorithm called self-correcting multi-channel Bussgang (SCMB) blind deconvolution algorithm was proposed. Unlike the original Bussgang blind deconvolution algorithm where the probability density function (pdf) of the signal being recovered is assumed to be completely known, the proposed SCMB blind deconvolution algorithm relaxes this restriction by parameterized the pdf with a Gaussian mixture model and expectation maximization (EM) algorithm, an iterative maximum likelihood approach, is employed to estimate the parameter side by side with the estimation of the equalization filters of the original Bussgang blind deconvolution algorithm. A feedback loop is also designed to compensate the effect of the parameter estimation error on the estimation of the equalization filters. Application of the SCMB blind deconvolution framework for binary image restoration, multi-pass synthetic aperture radar (SAR) autofocus and inverse synthetic aperture radar (ISAR) autofocus are exploited with great results.
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Law, Eugene. "HOW WELL DOES A BLIND, ADAPTIVE CMA EQUALIZER WORK IN A SIMULATED TELEMETRY MULTIPATH ENVIRONMENT." International Foundation for Telemetering, 2004. http://hdl.handle.net/10150/604926.
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International Telemetering Conference Proceedings / October 18-21, 2004 / Town & Country Resort, San Diego, CaliforniaThis paper will present the results of experiments to characterize the performance of a blind, adaptive constant modulus algorithm (CMA) equalizer in simulated telemetry multipath environments. The variables included modulation method, bit rate, received signal-to-noise ratio, delay of the indirect path relative to the direct path, amplitude of the indirect path relative to the direct path, and fade rate. The main measured parameter was bit error probability (BEP). The tests showed that the equalizer usually improved the data quality in the presence of multipath.
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Reinholdsen, Fredrik. "A Blind Constellation Agnostic VAE Channel Equalizer and Non Data-Assisted Synchronization." Thesis, Luleå tekniska universitet, Institutionen för system- och rymdteknik, 2021. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-86062.
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High performance and high bandwidth wireless digital communication underlies much of modern society. Due to its high value to society, new and improved digital communication technologies, allowing even higher speeds, better coverage, and lower latency are constantly being developed. The field of Machine Learning has exploded in recent years, showing incredible promise and performance at many tasks in a wide variety of fields. Channel Equalization and synchronization are critical parts of any wireless communication system, to ensure coherence between the transmitter and receiver, and to compensate for the often severe channel conditions. This study mainly explores the use of a Variational Autoencoder (VAE) architecture, presented in a previous study, for blind channel equalization without access to pilot symbols or ground-truth data. This thesis also presents a new, non data-assisted method of carrier frequency synchronization based around the k-means clustering algorithm. The main addition of this thesis however is a constellation agnostic implementation of the reference VAE architecture, for equalization of all rectangular QAM constellations. The approach significantly outperforms the traditional blind adaptive Constant Modulus algorithm (CMA) on all tested constellations and signal to noise ratios (SNRs), nearly equaling the performance of a non-blind Least Mean Squares (LMS) based Decision Feedback Equalizer (DFE).
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Lee, Yu-Min, and 李育民. "Blind Equalization for High Order QAM." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/58841307523330372725.
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碩士國立交通大學
電信研究所
81
In terrestial digital radio systems,the use of adaptive equalizers to combat with multipath distortion is desired. Conventional equalization techniques require that a training sequence known to the receiver is sent periodically such that the equalizer can determine the inverse channel transfer function via a deconvolution procedure. This kind of processing can be costly and may not be achievable in many applications such as multi-point networks. The desirability of avoiding the difficulty of the training sequence driven process prompts the development of a class of blind equalizers which achieve parameter estimation based on the channel output and prior knowledge about the input sequence. In this thesis,we propose a blind equalizer whose operation is based on the minimization of a noncovex cost function derived from the geometric and statistical properties of the QPSK constellation. Extension of the proposed method to QAM signals is straightforward with a minor modification of the cost function. Simulation results demonstrate that the proposed method achieves the performance of Godard's method for modreate channel conditions, and outperforms the latter for high order QAM signals. A merit of the proposed method not shared by Godard's is that it is immune to the random phase error incurred with the channel distortion. As a preliminary trial, We also incorporate spatial domain processing via an adaptive array to further improve the performance of the equalizer. The convergence property of the proposed algorithm is analyzed mathematically and confirmed by computer simulations.
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Lee, Hsiao-Fu, and 李孝甫. "IIR Lattice-based Blind Equalization Algorithms." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/87448900891724583497.
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碩士輔仁大學
電機工程學系
100
This thesis develops infinite impulse response (IIR) lattice structures based on two widely known blind equalization algorithms, which are the multi-modulus algorithm (MMA) and the constant modulus algorithm (CMA). The motivation for doing so is that generally for algorithms to achieve the same performance, an IIR filter requires fewer coefficients than its corresponding finite impulse response (FIR) counterpart. Moreover, the monitoring of the stability of an IIR lattice filter is simpler and less computationally complicated than that of the direct-form IIR filter. In computer simulations, the IIR lattice-based blind equalization algorithm was compared with the corresponding FIR transversal algorithms and FIR-based lattice algorithms. Simulation results demonstrate that the IIR lattice-based blind equalization algorithm is especially suitable for equalizing channels yielding inter-symbol interference (ISI) that mostly results from the post-cursor term.
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XIE, MING-ZHENG, and 謝鳴正. "Application of RLS algorithm in blind equalization." Thesis, 1992. http://ndltd.ncl.edu.tw/handle/27353553076924056175.
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Tsai, Chia-feng, and 蔡嘉峰. "The Blind Equalization Design in MC-CDMA System." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/61347349120232959761.
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Tseng, Hsien-Sheng, and 曾憲聖. "Blind Zero-Forcing Equalization Based on MMSE Criterion." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/68071510473903930432.
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碩士中原大學
電子工程研究所
92
Blind channel identification and equalization become important and popular in digital communication system. Since T. Kailath et al have proposed the algorithms utilizing the cyclostationary signal via oversampling the received signals or receiving signals by antenna arrays. A number of blind equalization methods based on SIMO system model and second order statistics of channel output have been developed. In this paper, we derive a new algorithm for blind equalization based on zero-forcing condition and minimum mean square error criterion. Minimizing the mean square error based on the different delay equalizers’ output . All possible delay zero-forcing equalizers can be computed simultaneously without channel coefficients matrix estimation so that the influence of the channel identification is reduced. Simulations are presented to demonstrate adequate performance of the new blind equalization algorithm.
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Liu, G. S., and 劉吉祥. "Linear Prediction Methods for Direct Blind MMSE Equalization." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/88904630936744020987.
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碩士國立交通大學
電信工程系
89
The channel equalization using the high-order statistics methods has a slow convergence rate. In recent years, the second-order statistics (SOS) methods have become a popular research. One of the SOS methods, such as the Ding algorithm proposed by Ding in 2000 is an advanced type of outer-product decomposition algorithm (OPDA), has been shown to have better performance than many existing algorithms. But Ding algorithm needs the pseudo-inverse of the correlation matrix, thus the computation is not simple and could cause numerical problems. It is also not suit for tracking time-varying channels. By the use of linear prediction (LP) method proposed by Fan, we deduce a new algorithm based on Ding algorithm. We name the new algorithm as linear prediction based outer-product decomposition algorithm (LP-OPDA). LP-OPDA combines both the advantages of LP and Ding algorithm and has its new advantages. LP-OPDA does not need the pseudo-inverse operation, thus have superior performance over Ding algorithm. LP-OPDA is available for tracking time-varying channels and also computationally efficient. From the simulation results, we can see LP-OPDA has superior performance over Ding algorithm and many other existing algorithms in many ways.
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Lee, Wen-Ta, and 李文哲. "Classification of voiceband digital signal using blind equalization." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/94370693333857437936.
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碩士大同工學院
電機工程學系
84
In this thesis, we have discussed the real time classification of high speed modem signals by using blind equalization. Several different self-recovering equalization techniques were investigated, in spite of some initial promising results, none of these other options worked better for the signal classification application than Godard*s algorithm. A classification scheme proposed by Benvenuto, which computes the probabilities that the unknown test signal belongs to each of several predetermined classes. We adopted Benvenuto*s algorithm applying CCITT modem signals , including the newest V.34 standard. The use of this approach resulted in an overall classification accuracy of about 97 % with detection time equals 1.675 sec from six classes of high speed modem signals.
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Chien, Wen-Chun, and 簡文君. "Convergence Analysis of MultiModulus-Based Blind Equalization Algorithms." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/85057441434838304919.
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碩士輔仁大學
電子工程學系
93
Unlike traditional channel equalizers, blind equalizers do not require a training sequence to start up or restart when the communication unexpectedly breaks down. This blind start-up ability is particularly useful in applications such as broadcast and point-to-multipoint networks. In this thesis, we propose a new blind equalization algorithm referred to herein as the multimodulus stop-and-go decision-directed algorithm (MSG-DDA) by combining the advantages of both an existing multimodulus algorithm (MMA) and a well-known stop-and-go algorithm (SGA). The dynamic convergence properties of the MSG-DDA, the MMA, and the SGA for blind equalization are mathematically analyzed using a conditional Gaussian approximation. The derived theoretical mean-squared-error (MSE) trajectories for the three algorithms are compared with their corresponding simulation results. The results verify that the MSG-DDA substantially outperforms both the MMA and SGA.
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HUANG, SEN-LONG, and 黃森隆. "Fast blind equalization using a block-updated algorithm." Thesis, 1991. http://ndltd.ncl.edu.tw/handle/04607976860670094687.
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Hsu, Chung-Ting, and 徐忠廷. "On the Improvement of Blind Channel Identification and Equalization." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/85000546920026282044.
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碩士國立臺灣大學
電信工程學研究所
89
Many channel identification and equalization techniques are used for reducing the ISI effect . Some improvement of the TXK blind channel identification technique are proposed in this thesis . At first, we introduced the TXK method and some kinds of modified method to overcome the finite sample effect and the phase shifting effect . After that, we modified these mothods to get better proformance and only increase less computational complexitity . Finally, we proposed a efficient method with cyclostationarity. It has much improvement of proformace in simulation .