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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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Mohanty, Nirode C. "ADAPTIVE EQUALIZER FOR M-ARY PSK MODULATION." International Foundation for Telemetering, 1985. http://hdl.handle.net/10150/615721.
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International Telemetering Conference Proceedings / October 28-31, 1985 / Riviera Hotel, Las Vegas, NevadaAn adaptive equalizer, based on a minimum mean square error criterion, has been derived for the purpose of extracting PSK signals transmitted through an unknown and asymmetric channel. The weights of the equalizer are obtained by using a simple formula containing the transform of the parallel channels. The performance of the equalizer is expressed in terms of the variance of the estimation error. The error is shown to be much less than that of the direct demodulated data.
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Roy, Tamoghna. "BER Modeling for Interference Canceling Adaptive NLMS Equalizer." Thesis, Virginia Tech, 2014. http://hdl.handle.net/10919/78055.
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Adaptive LMS equalizers are widely used in digital communication systems for their simplicity in implementation. Conventional adaptive filtering theory suggests the upper bound of the performance of such equalizer is determined by the performance of a Wiener filter of the same structure. However, in the presence of a narrowband interferer the performance of the LMS equalizer is better than that of its Wiener counterpart. This phenomenon, termed a non-Wiener effect, has been observed before and substantial work has been done in explaining the underlying reasons. In this work, we focus on the Bit Error Rate (BER) performance of LMS equalizers.
At first a model – the Gaussian Mixture (GM) model – is presented to estimate the BER performance of a Wiener filter operating in an environment dominated by a narrowband interferer. Simulation results show that the model predicts BER accurately for a wide range of SNR, ISR, and equalizer length. Next, a model similar to GM termed the Gaussian Mixture using Steady State Weights (GMSSW) model is proposed to model the BER behavior of the adaptive NLMS equalizer. Simulation results show unsatisfactory performance of the model. A detailed discussion is presented that points out the limitations of the GMSSW model, thereby providing some insight into the non-Wiener behavior of (N)LMS equalizers. An improved model, the Gaussian with Mean Square Error (GMSE), is then proposed. Simulation results show that the GMSE model is able to model the non-Wiener characteristics of the NLMS equalizer when the normalized step size is between 0 and 0.4. A brief discussion is provided on why the model is inaccurate for larger step sizes.Master of Science
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Wickert, Mark, Shaheen Samad, and Bryan Butler. "AN ADAPTIVE BASEBAND EQUALIZER FOR HIGH DATA RATE BANDLIMITED CHANNELS." International Foundation for Telemetering, 2006. http://hdl.handle.net/10150/604050.
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ITC/USA 2006 Conference Proceedings / The Forty-Second Annual International Telemetering Conference and Technical Exhibition / October 23-26, 2006 / Town and Country Resort & Convention Center, San Diego, CaliforniaMany satellite payloads require wide-band channels for transmission of large amounts of data to users on the ground. These channels typically have substantial distortions, including bandlimiting distortions and high power amplifier (HPA) nonlinearities that cause substantial degradation of bit error rate performance compared to additive white Gaussian noise (AWGN) scenarios. An adaptive equalization algorithm has been selected as the solution to improving bit error rate performance in the presence of these channel distortions. This paper describes the design and implementation of an adaptive baseband equalizer (ABBE) utilizing the latest FPGA technology. Implementation of the design was arrived at by first constructing a high fidelity channel simulation model, which incorporates worst-case signal impairments over the entire data link. All of the modem digital signal processing functions, including multirate carrier and symbol synchronization, are modeled, in addition to the adaptive complex baseband equalizer. Different feedback and feed-forward tap combinations are considered as part of the design optimization.
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Schumacher, Robert G. Jr. "An Efficient FPGA Implementation of a Constant Modulus Algorithm Equalizer for Wireless Telemetry." University of Dayton / OhioLINK, 2014. http://rave.ohiolink.edu/etdc/view?acc_num=dayton1417738709.
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Narayanan, Anand. "Eye opening monitor for optimized self-adaptation of low-power equalizers in multi-gigabit serial links." Thesis, Linköpings universitet, Elektroniksystem, 2013. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-106580.
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In modern day communication systems, there is a constant demand for increase in transmission rates. This is however limited by the bandwidth limitation of the channel. Inter symbol interference (ISI) imposes a great threat to increasing data rates by degrading the signal quality. Equalizers are used at the receiver to compensate for the losses in the channel and thereby greatly mitigate ISI. Further, an adaptive equalizer is desired which can be used over a channel whose response is unknown or is time-varying. A low power equalizing solution in a moderately attenuated channel is an analog peaking filter which boosts the signal high frequency components. Such conventional continuous time linear equalizers (CTLE) provide a single degree of controllability over the high frequency boost. A more complex CTLE has been designed which has two degrees of freedom by controlling the high frequency boost as well as the range of frequencies over which the boost is applied. This extra degree of controllability over the equalizer response is desired to better adapt to the varying channel response and result in an equalized signal with a wider eye opening. A robust adaptation technique is necessary to tune the equalizer characteristics. Some of the commonly used techniques for adaptation of CTLEs are based on energy comparison criterion in the frequency domain. But the adaptation achieved using these techniques might not be optimal especially for an equalizer with two degrees of controllability. In such cases an eye opening monitor (EOM) could be used which evaluates the actual signal quality in time domain. The EOM gives an estimate on the signal quality by measuring the eye opening of the equalized signal in horizontal and vertical domain. In this thesis work a CTLE with two degrees of freedom with an EOM based adaptation system has been implemented.
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Teekapakvisit, Chakree. "Low Complexity Adaptive Iterative Receivers for Layered Space-Time Coded and CDMA Systems." University of Sydney, 2007. http://hdl.handle.net/2123/1776.
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Doctor of Philosophy(PhD)In this thesis, we propose and investigate promising approaches for interference mitigation in multiple input multiple output (MIMO) and code division multiple access (CDMA) systems. Future wireless communication systems will have to achieve high spectral efficiencies in order to meet increasing demands for huge data rates in emerging Internet and multimedia services. Multiuser detection and space diversity techniques are the main principles, which enable efficient use of the available spectrum. The main limitation for the applicability of the techniques in these practical systems is the high complexity of the optimal receiver structures. The research emphasis in this thesis is on the design of a low complexity interference suppression/cancellation algorithm. The most important result of our research is the novel design of interference cancellation receivers which are adaptive and iterative and which are of low computational complexity. We propose various adaptive iterative receivers, based on a joint adaptive iterative detection and decoding algorithm. The proposed receiver can effectively suppress and cancel co-channel interference from the adjacent antennas in the MIMO system with a low computation complexity. The proposed adaptive detector, based on the adaptive least mean square (LMS) algorithm, is investigated and compared with the non-adaptive iterative receiver. Since the LMS algorithm has a slow convergence speed, a partially filtered gradient LMS (PFGLMS) algorithm, which has a faster convergence speed, is proposed to improve the convergence speed of the system. The performance and computational complexity of this receiver are also considered. To further reduce the computational complexity, we apply a frequency domain adaptation technique into the adaptive iterative receivers. The system performance and complexity are investigated. It shows that the computational complexity of the frequency domain based receiver is significantly lower than that of the time domain based receiver with the same system performance. We also consider applications of MIMO techniques in CDMA systems, called MIMO-CDMA. In the MIMO-CDMA, the presence of the co-channel interference (CCI) from the adjacent antennas and multiple access interference (MAI) from other users significantly degrades the system performance. We propose an adaptive iterative receiver, which provides the capability to effectively suppress the interference and cancel the CCI from the adjacent antennas and the MAI from other users so as to improve the system performance. The proposed receiver structure is also based on a joint adaptive detection and decoding scheme. The adaptive detection scheme employs an adaptive normalized LMS algorithm operating in the time and frequency domain. We have investigated and compared their system performance and complexity. Moreover, the system performance is evaluated by using a semi-analytical approach and compared with the simulation results. The results show that there is an excellent agreement between the two approaches.
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Teekapakvisit, Chakree. "Low Complexity Adaptive Iterative Receivers for Layered Space-Time Coded and CDMA Systems." Thesis, The University of Sydney, 2006. http://hdl.handle.net/2123/1776.
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In this thesis, we propose and investigate promising approaches for interference mitigation in multiple input multiple output (MIMO) and code division multiple access (CDMA) systems. Future wireless communication systems will have to achieve high spectral efficiencies in order to meet increasing demands for huge data rates in emerging Internet and multimedia services. Multiuser detection and space diversity techniques are the main principles, which enable efficient use of the available spectrum. The main limitation for the applicability of the techniques in these practical systems is the high complexity of the optimal receiver structures. The research emphasis in this thesis is on the design of a low complexity interference suppression/cancellation algorithm. The most important result of our research is the novel design of interference cancellation receivers which are adaptive and iterative and which are of low computational complexity. We propose various adaptive iterative receivers, based on a joint adaptive iterative detection and decoding algorithm. The proposed receiver can effectively suppress and cancel co-channel interference from the adjacent antennas in the MIMO system with a low computation complexity. The proposed adaptive detector, based on the adaptive least mean square (LMS) algorithm, is investigated and compared with the non-adaptive iterative receiver. Since the LMS algorithm has a slow convergence speed, a partially filtered gradient LMS (PFGLMS) algorithm, which has a faster convergence speed, is proposed to improve the convergence speed of the system. The performance and computational complexity of this receiver are also considered. To further reduce the computational complexity, we apply a frequency domain adaptation technique into the adaptive iterative receivers. The system performance and complexity are investigated. It shows that the computational complexity of the frequency domain based receiver is significantly lower than that of the time domain based receiver with the same system performance. We also consider applications of MIMO techniques in CDMA systems, called MIMO-CDMA. In the MIMO-CDMA, the presence of the co-channel interference (CCI) from the adjacent antennas and multiple access interference (MAI) from other users significantly degrades the system performance. We propose an adaptive iterative receiver, which provides the capability to effectively suppress the interference and cancel the CCI from the adjacent antennas and the MAI from other users so as to improve the system performance. The proposed receiver structure is also based on a joint adaptive detection and decoding scheme. The adaptive detection scheme employs an adaptive normalized LMS algorithm operating in the time and frequency domain. We have investigated and compared their system performance and complexity. Moreover, the system performance is evaluated by using a semi-analytical approach and compared with the simulation results. The results show that there is an excellent agreement between the two approaches.
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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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Howie, David Baird. "An investigation into non-linear propagation of MSK with a view to specifying an adaptive equalizer." Master's thesis, University of Cape Town, 1989. http://hdl.handle.net/11427/23250.
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The purpose of this dissertation is to investigate the effects of non-linear propagation on a proposed digital microwave radio link which employs MSK modulation, in order to specify a suitable form of adaptive equalization. MSK is a coherent modulation technique, having improved spectral roll-off over FSK because it avoids the abrupt phase changes at the bit transitions. However computer simulations and field results indicate that MSK digital radio links do suffer from intersymbol interference and crosstalk. Software and hardware simulations of multipath propagation are based on Rummlers simplified three path model and statistics. The results obtained from the computer simulations of the MSK link and multipath propagation confirm that there is no simple relationship between the multipath parameters and the BER degradation which could be used in the design of an equalizer. The choice of adaptive equalizer is made based on criteria such as construction cost, circuit complexity, and performance improvement. It is known from ray model analysis that at a transmitting frequency of 23 GHz deep fading will only occur on links longer than 5.24 kms. However even on hops of length 5 km' s the fade time is-in the order of 1612 seconds/month (calculated using Rummlers model). A ldB increase in theoretical Eb/No will also be required to overcome potential modem imperfections. It is necessary to have a time domain equalizer which can compensate for both amplitude and phase distortions simultaneously by acting directly on the ISI. The equalizer structure chosen is a 2-by-2, fractionally spaced, decision feedback, complex adaptive equalizer with zero forcing control algorithm.
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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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Roy, Tamoghna. "Non-Wiener Characteristics of LMS Adaptive Equalizers: A Bit Error Rate Perspective." Diss., Virginia Tech, 2018. http://hdl.handle.net/10919/92869.
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Adaptive Least Mean Square (LMS) equalizers are widely used in digital communication systems primarily for their ease of implementation and lack of dependence on a priori knowledge of input signal statistics. LMS equalizers exhibit non-Wiener characteristics in the presence of a strong narrowband interference and can outperform the optimal Wiener equalizer in terms of both mean square error (MSE) and bit error rate (BER). There has been significant work in the past related to the analysis of the non-Wiener characteristics of the LMS equalizer, which includes the discovery of the shift in the mean of the LMS weights from the corresponding Wiener weights and the modeling of steady state MSE performance. BER performance is ultimately a more practically relevant metric than MSE for characterizing system performance. The present work focuses on modeling the steady state BER performance of the normalized LMS (NLMS) equalizer operating in the presence of a strong narrowband interference.
Initial observations showed that a 2 dB improvement in MSE may result in two orders of magnitude improvement in BER. However, some differences in the MSE and BER behavior of the NLMS equalizer were also seen, most notably the significant dependence (one order of magnitude variation) of the BER behavior on the interference frequency, a dependence not seen in MSE. Thus, MSE cannot be used as a predictor for the BER performance; the latter further motivates the pursuit of a separate BER model.
The primary contribution of this work is the derivation of the probability density of the output of the NLMS equalizer conditioned on a particular symbol having been transmitted, which can then be leveraged to predict its BER performance. The analysis of the NLMS equalizer, operating in a strong narrowband interference environment, resulted in a conditional probability density function in the form of a Gaussian Sum Mixture (GSM). Simulation results verify the efficacy of the GSM expression for a wide range of system parameters, such as signal-to-noise ratio (SNR), interference-to-signal (ISR) ratio, interference frequency, and step-sizes over the range of mean-square stable operation of NLMS. Additionally, a low complexity approximate version of the GSM model is also derived and can be used to give a conservative lower bound on BER performance.
A thorough analysis of the MSE and BER behavior of the Bi-scale NLMS equalizer (BNLMS), a variant of the NLMS equalizer, constitutes another important contribution of this work. Prior results indicated a 2 dB MSE improvement of BNLMS over NLMS in the presence of a strong narrowband interference. A closed form MSE model is derived for the BLMS algorithm. Additionally, BNLMS BER behavior was studied and showed the potential of two orders of magnitude improvement over NLMS. Analysis led to a BER model in the form of a GSM similar to the NLMS case but with different parameters. Simulation results verified that both models for MSE and BER provided accurate prediction of system performance for different combinations of SNR, ISR, interference frequency, and step-size.
An enhanced GSM (EGSM) model to predict the BER performance for the NLMS equalizer is also introduced, specifically to address certain cases (low ISR cases) where the original GSM expression (derived for high ISR) was less accurate. Simulation results show that the EGSM model is more accurate in the low ISR region than the GSM expression. For the situations where the derived GSM expression was accurate, the BER estimates provided by the heuristic EGSM model coincided with those computed from the GSM expression.
Finally, the two-interferer problem is introduced, where NLMS equalizer performance is studied in the presence of two narrowband interferers. Initial results show the presence of non-Wiener characteristics for the two-interferer case. Additionally, experimental results indicate that the BER performance of the NLMS equalizer operating in the presence of a single narrowband interferer may be improved by purposeful injection of a second narrowband interferer.PHD
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Gao, Wei, Shih-Ho Wang, and Kamilo Feher. "TESTS AND EVALUATIONS OF ADAPTIVE FEHER EQUALIZERS FOR A LARGE CLASS OF SYSTEMS, INCLUDING FQPSK." International Foundation for Telemetering, 2000. http://hdl.handle.net/10150/606772.
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International Telemetering Conference Proceedings / October 23-26, 2000 / Town & Country Hotel and Conference Center, San Diego, CaliforniaDesign and performance evaluation of a low-complexity equalizer for recently standardized spectral efficient Feher patented quadrature phase shift keying (FQPSK) system [1] over multipath fading channel is presented. The implementation based on a Feher patented equalizer (FE) [1] is of a structure with three branches, which are individually used to compensate for a moving fade notch with different locations. These branches are switched by the control signal that is generated based on pseudo-error on-line detection technique. It is demonstrated that for typical aeronautical telemetry RF frequency selective fading channels, having delay spreads in 20 – 200 ns range, the adaptive FE reduces the number of statistical outages by more than 60% without the need for training bits and without increasing the receiver synchronization time.
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Krejča, Libor. "Implementace algoritmů ekvalizace přenosového kanálu v FMT modulaci." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2010. http://www.nusl.cz/ntk/nusl-218254.
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The objective of Diploma thesis is design of analysis tool for equalizers used in FMT modulation. The model of transmission channel was designed for simulations with FMT. The transmission path is modeled by test loop, which corresponds to DSL line. For this reason, some principles of DSL technology is described in the thesis. The principles of multicarrier modulation are introduced in first part. The multicarrier modulation with filter bank (FMT) is described in detail.The different methods of design the fiter bank are given and compared. Channel equalization are introduced in second part. The attention was focused on minimum mean square error filtering (MMSE). Decision feedback channel equalizer (DFE) is extended from linear MMSE equalizer. DFE equalizers were programmed in analysis tool. For computation of equalizer coeficients was used also equalizers based on adaptive algorithms and MMSE. The last part describes the results of DFE equalizers used in communication system with FMT modulation. Analysis tool was programmed in MATLAB with a graphical user interface. It allows to show mean square error, signal-to-noise ratio and transmission speed dependence on delay between original and distorted signal. Signal-to-noise ration is displayed also in individual subchannels and users can display mean square error dependence on different orders of DFE filters.
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Guzeev, Andrew. "Use of equalization and echo canceling on circuit board wires." Thesis, Linköping University, Department of Electrical Engineering, 2002. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1466.
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Advances in CMOS technology have resulted in increased clock fre-quencies, even exceeding 3GHz. At the same time, frequencies on most board wires are 125-800MHz. It is especially problematic in modern computer mem-ory buses and high speed telecommunication devices, such as switches and routers operating at 10Gb/s on its ports. It is believed that circuit board buses can be used up to about 20GHz, but there is a problem with Intersymbol Inter-ference (ISI) causing distortion of transmitted symbols by multiple reflections.
Actually, the circuit board bus behaves like a passive low pass filter with unknown (perhaps changing) transfer characteristic. The problem of ISI was solved some time ago in the telecommunication area. With use of adaptive equalizers it is possible to increase throughput of a long distance communication channel dramatically.
But the microprocessor bus has certain differences from telecommunica-tion devices such as modems. First of all, the clock frequency on a bus is much higher than in modems. Secondly, a bus has a much more complex structure than a telecommunication channel. At the same time, we can’t use a lot of re-sources for bus maintaining.
The aim of the thesis work is to investigate the possibility of using adap-tive equalization on a bus, and the construction of a reasonable mathematical model of such an equalizer. Also limits of equalizationare examined and de-pendencies are derived.
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Dick, Chris. "FPGAs: RE-INVENTING THE SIGNAL PROCESSOR." International Foundation for Telemetering, 2002. http://hdl.handle.net/10150/606348.
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International Telemetering Conference Proceedings / October 21, 2002 / Town & Country Hotel and Conference Center, San Diego, CaliforniaFPGAs are increasingly being employed for building real-time signal processing systems. They have been used extensively for implementing the PHY in software radio architectures. This paper provides a technology and market perspective on the use FPGAs for signal processing and demonstrates FPGA DSP using an adaptive channel equalizer case study.
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Sousa, Tiago Fernando Barbosa de. "Equaliza??o neural aplicada a sistemas com modula??o bidimensional em fibra ?ptica." Universidade Federal do Rio Grande do Norte, 2014. http://repositorio.ufrn.br:8080/jspui/handle/123456789/15498.
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Nowadays, optic fiber is one of the most used communication methods, mainly due to the fact that the data transmission rates of those systems exceed all of the other means of digital communication. Despite the great advantage, there are problems that prevent full utilization of the optical channel: by increasing the transmission speed and the distances involved, the data is subjected to non-linear inter symbolic interference caused by the dispersion phenomena in the fiber. Adaptive equalizers can be used to solve this problem, they compensate non-ideal responses of the channel in order to restore the signal that was transmitted. This work proposes an equalizer based on artificial neural networks and evaluates its performance in optical communication systems. The proposal is validated through a simulated optic channel and the comparison with other adaptive equalization techniques
A fibra ?ptica ? um dos meios de comunica??o mais utilizados atualmente, principalmente devido ao fato da taxa de transmiss?o de dados desses sistemas excederem as de todos os outros meios de comunica??o digital. Apesar desta grande vantagem, existem problemas que impedem o total aproveitamento do canal ?ptico: com o aumento da velocidade de transmiss?o e das dist?ncias envolvidas, os dados ficam sujeitos a interfer?ncia intersimb?lica n?o linear, causada pelos fen?menos de dispers?o na fibra ?ptica. Para solucionar esse problema podem ser utilizados equalizadores adaptativos, que compensam respostas n?o ideais do canal, com o intuito de restaurar o sinal que foi transmitido. Neste trabalho apresentamos uma proposta de equalizador baseado em redes neurais artificiais e avaliamos seu desempenho em sistemas de comunica??o ?ptica. A proposta ? validada em um canal ?ptico simulado e comparada a outras t?cnicas de equaliza??o adaptativa
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Shahid, Muhammad. "Methods for Objective and Subjective Video Quality Assessment and for Speech Enhancement." Doctoral thesis, Blekinge Tekniska Högskola [bth.se], Faculty of Engineering - Department of Applied Signal Processing, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00603.
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The overwhelming trend of the usage of multimedia services has raised the consumers' awareness about quality. Both service providers and consumers are interested in the delivered level of perceptual quality. The perceptual quality of an original video signal can get degraded due to compression and due to its transmission over a lossy network. Video quality assessment (VQA) has to be performed in order to gauge the level of video quality. Generally, it can be performed by following subjective methods, where a panel of humans judges the quality of video, or by using objective methods, where a computational model yields an estimate of the quality. Objective methods and specifically No-Reference (NR) or Reduced-Reference (RR) methods are preferable because they are practical for implementation in real-time scenarios. This doctoral thesis begins with a review of existing approaches proposed in the area of NR image and video quality assessment. In the review, recently proposed methods of visual quality assessment are classified into three categories. This is followed by the chapters related to the description of studies on the development of NR and RR methods as well as on conducting subjective experiments of VQA. In the case of NR methods, the required features are extracted from the coded bitstream of a video, and in the case of RR methods additional pixel-based information is used. Specifically, NR methods are developed with the help of suitable techniques of regression using artificial neural networks and least-squares support vector machines. Subsequently, in a later study, linear regression techniques are used to elaborate the interpretability of NR and RR models with respect to the selection of perceptually significant features. The presented studies on subjective experiments are performed using laboratory based and crowdsourcing platforms. In the laboratory based experiments, the focus has been on using standardized methods in order to generate datasets that can be used to validate objective methods of VQA. The subjective experiments performed through crowdsourcing relate to the investigation of non-standard methods in order to determine perceptual preference of various adaptation scenarios in the context of adaptive streaming of high-definition videos. Lastly, the use of adaptive gain equalizer in the modulation frequency domain for speech enhancement has been examined. To this end, two methods of demodulating speech signals namely spectral center of gravity carrier estimation and convex optimization have been studied.
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Žlebek, Lukáš. "Ekvalizace přenosového kanálu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2018. http://www.nusl.cz/ntk/nusl-377143.
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This thesis describes a design of a simulation of transmission of digital information via communication system and equalization of communication function. The layout of communication channel with multiway transmission is described in following part. Next part is about hardware modulator which generate modulated signal which is transmitted via communication channel and after is sampled by A/D convertion card to computer, where is equalizated and demodulated in Simulink. In the last part of this thesis, there is proposal of the laboratory task and its sample solution.
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Golestaneh, Shahram Carleton University Dissertation Engineering Electrical. "Adjacent channel interference reduction by adaptive equalizers." Ottawa, 1992.
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Wong, Choong Hin. "Wideband adaptive full response multilevel transceivers and equalizers." Thesis, University of Southampton, 1999. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.310547.
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蔡俊松. "Performance analysis of adaptive equalizer & combined adaptive equalizer for channel equalization." Thesis, 1987. http://ndltd.ncl.edu.tw/handle/76484211638344319072.
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Eng, Bor-Yang, and 曾柏元. "Adaptive Linear Equalizer Generator." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/44219316593454235802.
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碩士華梵大學
電子工程學系碩士班
94
Intersymbol interference(ISI) is an important factor which affects the performance of communication systems.To achieve a reliable digital communication, an equalizer can effectively lower ISI caused by a band-limited or a multipath channel. In this thesis,we present a high-level design flow for an adaptive linear Equalizer(EQ).With the length of the EQ and the word length of the input signal,an adaptive linear EQ Verilog code can be generated by the proposed MATLAB program. The resultant EQ verilog code can be verified on the ModelSim simulator. It can be optimally prototyped on the Altera CycloneⅡ Development Board as well.
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Chang, Chih-Chiang, and 張志強. "Design of Blind Adaptive Equalizer." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/83036786785594346643.
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碩士國立臺灣海洋大學
電機工程學系
92
In wireless communication systems, equalizer is needed to suppress the intersymbol interference (ISI) caused by the channel effect of multipath propagation. The purpose of blind equalizer is to correctly estimate the transmitted message sequences directly from the received signal without the help of training sequences. In this paper, we propose a novel method, that exploits the cyclostationary property of communication signals and uses the phase-locked loop, to design a blind adaptive equalizer. The Quadrature Phase Shift Keying (QPSK) modulated signal with pulse shaping obtained by the raise cosine filter is used as the transmitting signal. The proposed blind adaptive equalizer uses the least-mean-square (LMS) and recursive least-squares (RLS) algorithms to minimize the error signal between the transformed feed forward filter output signal and a complex exponential signal extracted from the phase-locked loop as well as a LMS-based feedback filter for decision-feedback equalization. Because the proposed blind adaptive equalizer can perform equalization without using training sequences, it exhibits faster convergence speed and better bit-error rate at high signal-to-noise ratios than conventional equalizers which need usage of training sequences, thus achieving higher spectral efficiency. Therefore, the proposed equalizer can improve communi- cation quality and increase the capacity. To evaluate the performance of the proposed equalizer, computer simulations were carried out for the cases of stationary and nonstationary channels. Keywords：equalizer, blind, adaptive, cyclostationary, spectral line, LMS algorithm, RLS algorithm.
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Chen, Kuan-Yu, and 陳冠宇. "Design of 20Gbps Adaptive Linear Equalizer and Decision Feedback Equalizer." Thesis, 2016. http://ndltd.ncl.edu.tw/handle/rc2864.
Full textAbstract:
碩士國立臺灣大學
電子工程學研究所
105
Nowadays, the SerDes (Serializer-to-Deserializer) topology is increasingly popular in the wireline communication systems for the reduced I/O pads and also the low fabrication cost. However, the aggregate bandwidth of the data traffic is strictly limited by the channel characteristics. The limited bandwidth of the channel will induce large inter-symbol interference (ISI), and also deteriorate the bit-error-rate (BER) performance. Thus, the equalization is more and more important in the wireline systems. Moreover, the channel attenuation greatly varies with materials and lengths, and hence the adaptation techniques for the equalizer are required in most applications. In this thesis, the most common equalizers in the receiver are designed, analyzed, and verified. The first part shows a 20Gbps linear equalizer with the proposed adaptation method. Fabricated in 40nm CMOS technology, this adaptive linear equalizer can well compensate the channel loss under 18.3dB attenuation. Only 2.68us is required for the adaptation procedure and 4.9mW is consumed by the adaptation logics. The second part presents a 20Gbps infinite impulse response decision feedback equalizer (IIR-DFE). To enhance the power efficiency of the IIR-DFE, the charge-steering logic (CSL) is utilized in this work. Besides, the quarter-rate topology and some circuit merging techniques are adopted. Fabricated in 40nm CMOS technology, the power efficiency of 0.31mW/Gbps can be obtained.
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Lin, Yuan-Fu, and 林元莆. "5~20 Gb/s Adaptive Linear Equalizer and Decision-Feedback Equalizer." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/13812443680406086957.
Full textAbstract:
碩士國立臺灣大學
電子工程學研究所
102
In recent years, in addition to the fast growing in data rate, wide-range data is also required in the applications of various multimedias and portable devices. As the data rate keeps rising, many significant problems appear. One is that the bandwidth is limited compared to the data rate. It will result a significant inter symbol interference (ISI) to degrade the bit error rate (BER). In order to deal with ISI, equalizers are widely adopted. However, the length or the material of the communication channel may be different depending on the application. Therefore, an adaptive algorithm with the equalizer is more popular in recent communication systems. In wide-range data rate application, power efficiency issue is also concerned. This thesis is mainly divided into two parts. In Chapter 2, a 5-20 Gb/s power scalable adaptive continuous-time linear equalizer (CTLE) architecture is proposed. We use a power scalable technique to improve the power efficiency for slow data rate. We also propose an adaptive algorithm using edge counting. This circuit is implemented in 40-nm CMOS process. A 5-20 Gb/s adaptive charge-steering decision-feedback equalizer (DFE) is presented in chapter 3. To lower power consumption of the system, charge-steering logic circuit is adopted in this design. We use sign-sign least mean square (SSLMS) algorithm to adjust the DFE’s taps adaptively. This circuit is implemented in 40-nm CMOS process.
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Yang, Gu Ming, and 顧明陽. "Lp norm backpropagation for adaptive equalizer." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/64225443979136227447.
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Lin, Wen-Hsin, and 林文信. "Adaptive Third-Order Volterra Satellite Channel Equalizer." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/41271981507198781753.
Full textAbstract:
碩士國立中山大學
電機工程學系研究所
89
Digital satellite communication systems are equipped with nonlinear amplifiers such as travelling wave tube (TWT) amplifiers at or near saturation for better efficiency. The TWT exhibits nonlinear distortion in both amplitude and phase (AM/AM and AM/PM) conversion, respectively. That is, in the digital satellite communication the transmission is disturbed not only by the non-linearity of transmitter amplifier, but also by the inter-symbol interference (ISI) with additive white Gaussian noise. To compensate the non-linearity of the transmitter amplifier and ISI, in this thesis, a new nonlinear compensation scheme consists of the predistorter and adaptive third-order Volterra-based equalizer, with the inverse QRD-RLS (IQRD-RLS) algorithm, which are located before and after the nonlinear channel, is proposed respectively. The third-order Volterra filter (TVF) equalizer based on the IQRD-RLS algorithm achieve superior performance, in terms of convergence rate, steady-state mean-squared error (MSE), and numerically stable. They are highly amenable to parallel implementation using array architectures, such as systolic arrays. The computer simulation results using the M-ary PSK modulation scheme are carried out the signal’s constellation diagrams, the learning curve of the MSE and the bit error rate (BER) are compared with conventional least mean square (LMS), gradient adaptive lattice (GAL) and adaptive LMS with lattice pre-filter algorithms.
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Chen, Kuang-Ren, and 陳光仁. "Adaptive Cable Equalizer Using Phase Detection Technique." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/78164516195620513499.
Full textAbstract:
碩士國立交通大學
電子研究所
100
The thesis presents two adaptive cable equalizers with novel time-domain approach called the phase detection technique for application in wireline communication. This detection mechanism avoids offset-sensitive rectifiers which normally exist in conventional equalizers. Edge-speed information is converted into phase information in this technique. The proposed detection mechanism is similar to digital control, which makes it more reliable and more immune to PVT variations; meanwhile, it has advantages of input swing and data pattern independent. In order to improve the bandwidth of equalizer filter, negative capacitance, inductive peaking and other skills are employed. Two chips are implemented in this thesis. Both of them adopt the phase detection technique. The first chip implements a 10Gb/s adaptive cable equalizer in TSMC 0.13μm CMOS technology. It can compensate a 24-inch channel on an FR-4 PCB, which has an 18dB loss at 5GHz. The power dissipation is 39mW excluding the output buffer from a 1.5-V supply voltage and the measured bit error rate is less than 10^-13. The second chip implements a 6Gb/s adaptive cable equalizer in TSMC 0.18μm CMOS technology. It can compensate a 61-inch channel on an FR-4 PCB, which has a 21dB channel loss at 3GHz. The power consumption is 31mW without the output buffer from a 1.8-V supply voltage and the measured bit error rate is less than 10^-13.
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Shayan, Shahramian. "A Pattern-guided Adaptive Equalizer in 65nm CMOS." Thesis, 2011. http://hdl.handle.net/1807/29618.
Full textAbstract:
This thesis presents the design, implementation, and fabrication of a pattern-guided equalizer in a 65nm CMOS process. By counting the occurrence of 6 out of 16 4-bit
patterns in the received data and utilizing their spectral content, the signal is equalized separately at fN and fN/2, where fN is half the bit rate. The design was packaged using a 64 pin Quad Flat No leads (QFN) package. Two different channels were used and the equalizer was able to open the eye for both 13dB and 17dB of attenuation at the Nyquist frequency. The adaptation performance was determined by measuring the vertical and horizontal eye openings for all possible equalizer coefficients. Measured results at 6Gb/s confirm that the adaptation engine opens a closed eye to within 2.6% of optimal vertical opening and 7% of optimal horizontal eye opening while consuming 16.8mW from a 1.2V supply.
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Chen, Ren-Zi, and 陳仁智. "Design Adaptive Equalizer on OFDM-CDMA Communication Systems." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/61131954334405389885.
Full textAbstract:
碩士國立清華大學
電機工程學系
87
A new multiple access communication system with orthogonal multicarrier (OM) transmission via Direct Sequence Spreading Spectrum (DS/SS) which can transmit data at high speed in wireless environmenthas attracted in recent years. The combination of the orthogonal frequency division multicarrier and Direct Sequence Code Division multiple access (OFDM--CDMA) provides an efficient and flexible way to access and distribute multiple multiplexed data source, and the OFDM--CDMA scheme proves to be robust in combating multipath and fading of transmission channels. In this paper, a new adaptive equalizer scheme for the receiver of OFDM--CDMA system is proposed to reduce the multiple access interference (MAI) arisen from multipath propagation and cross-correlation between the code sequences. The proposed adaptive equalizer consists of a cascade of an adaptive prefilter and an adaptive postfilter to take a balance between the compensation of channel effect and the MAI cancellation to avoid noise enhancement at the same time. We also propose new adaptive algorithm for blind channel estimation. Finally, the performance of the proposed method is analyzed and the asymptotic convergence property is also discussed. From several simulation results, we find that the proposed adaptive equalizer algorithm performs better than the other equalizers.
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LAI, KUN-CAI, and 賴坤財. "A decision-aided adaptive equalizer with simplified implementation." Thesis, 1987. http://ndltd.ncl.edu.tw/handle/76901894192424203090.
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Chung, Pei-Ju, and 鍾沛如. "Adaptive Time Domain Equalizer (TEQ) for VDSL System." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/cv4pcm.
Full textAbstract:
碩士國立交通大學
電機與控制工程系所
93
There has been great interest in DMT systems (Discrete Multitone) for high-speed transmission. The time domain equalizer (TEQ) plays an important role in such an application. The VDSL (Very-high-bit-rate Digital Subscriber Line)is an example of DMT systems. In this thesis, two adaptive TEQ design methods will be proposed. The proposed methods utilize training symbols in the initialization stage. We use an adaptive approach to train TEQ by exploiting the symbols in frequency domain. The simulation results will be given to illustrate the proposed TEQ methods can achieve good bit rates with only a small number of training symbols (iterations).
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Hung, Chia-Tse, and 洪嘉澤. "A 40 Gb/s PAM-4 Adaptive Equalizer." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/9ad436.
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Qui-Ting, Chen. "A 4-PAM Adaptive Analog Equalizer for Backplane Interconnections." 2006. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-2807200611502200.
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魏鴻富. "Adaptive equalizer for fast time-varying multipath fading channels." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/22885189023963158769.
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李佳勳. "Hardware Architecture Design of Adaptive Equalizer and TCM Decoder." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/rav63b.
Full textAbstract:
碩士國立交通大學
電信工程系所
93
The Single-pair High Speed Digital Subscriber Loop (SHDSL) is the new generation symmetric DSL technique which could supply at most 2.3 Mbps downlink and uplink data rate symmetrically to subscribers on a long loop. However, the InterSymbol Interference (ISI) is severe especially duration the transmission over a long loop. Normal data transmission is impossible without properly taking care of the ISI problem. To assure SHDSL tranceivers to provide full-rate transmission, we need a powerful adaptive equalizer to ease the ISI problem. The decision feedback equalizer is the most often used equalizer for sovling the ISI problem. However, it has the error propagation problem, which will degrade the system performance. To improve the performance, the joint equalization and channel decoding is necessary. Nevertheless, combining the trellis decoder in the decision feedback equalizer will result in high complexity hardware. A powerful equalizer called Tomlinson-Harashima precoder (THP) system was proposed to solve this problem. By use of the procoding technique, the joint equalization and channel decoding can be accomplished by cascade a linear equalizaer and a TCM decoder for channel coding. In this theisis, starting from algorithm design and computer simulation, we design the THP system and TCM decoder hardware architectures according to the G.SHDSL recommendation. The resulting hardware could achieve the maximum 2.3 Mbps data rate under the 50 MHz operation clock. The hardware was verified on the FPGA development board.
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Chen, Qui-Ting, and 陳坵鋌. "A 4-PAM Adaptive Analog Equalizer for Backplane Interconnections." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/89205174450270204222.
Full textAbstract:
碩士國立臺灣大學
電子工程學研究所
94
The exploring increasing of data rate has created a major challenge for electronic circuits used at the interface of the backplane physical layer links. As the data rate increases above Gb/s, intersymbol interference (ISI) becomes an essential issue in digital communications, limiting the achievable transmission speed and distance over channels. As to electronic compensation for the channel loss, digital or analog equalizers can be used. Digital (DSP based) equalization offers more accurate and higher performance comparing with analog counterpart. But the design of digital equalization has a bottleneck on the implementation of high-speed ADCs, which need large area and high power consumption. Consequently, pure analog equalizer is a more efficient solution. In this thesis, a 4-PAM (pulse amplitude modulation) adaptive analog equalizer is proposed to compensate the FR-4 PCB backplane interconnections by using a sum-feedback filter (SFF), relaxing the design requirement of the conventional analog feed-forward equalizers (FFE). 4-PAM is also adopted to increase the transmission data rate over bandwidth-limited channel. Fabricated in a standard 0.18-μm CMOS technology, the analog equalizer can successfully recover the 14 Gb/s random data transmitted over 40-inch copper channels while dissipating 121 mW from a 1.8-V power supply. The die size is 1.285 × 0.98 mm2.
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Chen, Hung-I., and 陳弘易. "The Design and Realization of Adaptive Coaxial Cable Equalizer." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/23493973469435968381.
Full textAbstract:
碩士國立臺灣大學
電機工程學研究所
87
The data transmission of high-speed, long-distance coaxial cable suffers magnitude losses due to its effective impedance. The loss is a function of signal frequency. To compensate the loss, we use a equalizer to rebuild the signal and increase its reliability. In this thesis, we design a equalizer for the application of the transmission line Belden 8281. It can also compensate various lengths of transmission lines via the control of a feedback loop. The data bit rate is 200Mbps, and the longest length of transmission line can be compensated is 125 meters. Besides the voltage-mode equalizer circuit proposed in the literature, we also proposed the current-mode equalizer circuit for realizing the wider-band performance and decreasing the jitter of the equalizer output signals. The prototype equalizer circuit has been fabricated in a double-poly double metal (DPDM) 0.35um CMOS technology.
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Maw-Lin, Leou, and 柳茂林. "Hybrid of Adaptive Array and Equalizer for Mobile Communications." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/53480715832960526580.
Full textAbstract:
博士國立臺灣大學
電機工程學研究所
87
The combination of adaptive array and equalizer (AE) has been developed for suppressing the co-channel interferers and the multipath interferers in mobile communications. In this dissertation a novel hybrid of the adaptive array and equalizer (NHAE) system is proposed to combat the problems of insufficient degrees of freedom and mainbeam multipath interferers. The NHAE utilizes a modified training sequence to adjust the weight vector of the array that leads the array to cancel only the co-channel interferers and the multipath interferers are removed by the equalizer following the array. Therefore, the array in the NHAE may need less number of the elements than the conventional array, which cancels both the co-channel interferers and multipath interferers. Besides, the presence of the mainbeam multipath interferers, which may seriously degrade the performance of the AE, has less effect on the NHAE since it is suppressed by the equalizer instead of the array. Simulation results are presented to demonstrate the merits of the NHAE. The NHAE has been proposed to combat both the co-channel interferers and multipath interferers in mobile communications under the training sequence is assumed available in the system. When the training sequence is absent, we propose a novel constant modulus algorithm (CMA) for the hybrid of adaptive array and equalizer (HAE) to combat the problems of the insufficient degrees of freedom and the mainbeam multipath interferers. In the system of the HAE with CMA, we exploit the constant modulus property of the output signal of the HAE to train the array and equalizer simultaneously. Therefore, the co-channel interferers can be canceled by the array and the multipath interferers can be removed by the array or by the equalizer following the adaptive array. Simulation results will prove that the problems of the insufficient degrees of freedom and mainbeam multipath interferers are solved by the HAE with novel CMA in case of training sequence is absent. Though the HAE with CMA can be used in the absence of training sequence, the convergent rate of the system is too slow for mobile communications. In this dissertation, we proposed a novel LMS for HAE to increase the convergent rate. The HAE with novel LMS utilizes the mean square error defined as the difference between the training sequence and the output signal of the HAE to adjust the weight vectors of the array and the equalizer simultaneously. Thus, the interferers, which include the co-channel interferers and the multipath interferers, can be canceled by not only the array but also the equalizer in the HAE with novel LMS. Simulation results show that the performances and convergent rate of the HAE with CMA can be improved.
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Hsiao, Jui-Cheng, and 蕭瑞成. "A 10Gb/s Equalizer with a Digital Adaptive Algorithm." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/79295111398802343972.
Full textAbstract:
碩士國立臺灣大學
電子工程學研究所
102
An equalizer using a digital adaptive algorithm is proposed to replace RC filters, rectifiers, V/I converters and large loop capacitors. The proposed algorithm uses two analog reference levels to represent the high-frequency and low-frequency components of the input data, respectively. By monitoring the two reference levels, the proposed equalizer can tune its high-frequency gain to compensate the channel loss appropriately. The measured peak-to-peak jitter for the 90-cm channel is reduced from 81.63 ps to 33.84 ps with 8-Gb/s input data; while the 37-cm channel is decreased from 47.64 ps to 37.60 ps with 10-Gb/s input data. The proposed adaptive equalizer successfully operates for different channel lengths (up to 90-cm) on FR-4 PCB. This work has been fabricated in a 40-nm process, and the equalizer core circuit occupies 0.014 mm2 and consumes 10 mW from a 1-V supply (excluding output buffer circuits). The proposed equalizer uses only digital logic gates for adaptation instead of RC filters, rectifiers, V/I converters and large capacitors for loop stabilization. Such digital-intensive implementation can highly reduce the hardware cost in advanced CMOS technologies.
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Lin, Min-chieh, and 林旻頡. "Adaptive Equalizer Analysis and Design Based on Volterra Series." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/75817977988036236003.
Full textAbstract:
碩士逢甲大學
資訊電機工程碩士在職專班
96
Adaptive equalizer has been widely employed in the fields of communication and automatic control. Among various types of adaptive algorithms, the normalized least mean square (NLMS) Volterra adaptive equalizer is proposed in this thesis. The block formulation of least mean square (LMS) adaptive Volterra equalizer is presented and this formulation has a mathematical equivalence with time domain sample processing LMS. Hence, it maintains the same performance while allowing a reduction in arithmetical complexity (even for small block size). The fast Newton type algorithm equipped with a second-order Volterra equalizer is developed, so as to detect fast ethernet network channel. The simulation results confirm the validity and performance of the advocated design methodology.
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Huang, Ying-Che, and 黃英哲. "Implementation of Blind Adaptive Equalizer based on DSP chip." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/88540880171352632449.
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碩士國立臺灣海洋大學
電機工程學系
93
When propagation through a underwater channel, acoustic waves are usually affected by reflection and scattering of underwater environment, therefore, signals are seriously distorted by multipath interference, leading to the intersymbol interference. Equalizer is one of the solutions to this problem. In this paper, we first investigate two blind adaptive equalizers, namely the blind constant modulus(CMA) and the PLL-based blind adaptive equalizer, and then compare their performances by computer simulations. We find that these two equalizer are essentially equivalent. For real-time communications, DSP chips are used for the firmware design and implementation of QPSK modulation and PLL-based blind equalization. Using the cyclostationarity of communication signals, the PLL-based equalizer does not require training sequence to adapt channel variations, exhibits faster convergence speed and better bit-error rate. For performance evaluation, the DSP-based transceiver is tested by data transmission through the channel of a watertank. Experimental results show that the transceiver can significantly improve the multipath effect. Keywords：underwater channel, channel distortion, equalizer, cyclostationary, blind, adaptive.
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Liao, Sheng-Hui, and 廖盛惠. "Soft Decision Approaches for Blind Adaptive Decision Feedback Equalizer." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/60781162149955816270.
Full textAbstract:
碩士國立臺灣海洋大學
電機工程學系
93
Because of multi-path and Doppler effect, the development of the underwater acoustic communication system is a challenging task. The optimum equalization and appropriate signal processing technique are needed to improve the quality of communications. In this thesis, the implementation algorithms based on the blind adaptive decision feedback equalization (BADFE), whose function can be automatically adjusted according to variations of channel, are investigated. The soft decision is combined with the BADFE to get a modified equalization scheme. The proposed method combines decision feedback, blind adaptive scheme, and the soft decision method at the same time. Therefore it can combat the distortion occurred in underwater communication channels without the need of the training sequence. The bit error rates and the transmission efficiency are improved. Computer simulation results indicate that the performances of the proposed method always has better convergence speed and lower bit error rate even in the bad channel and low SNR.
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SHI, JUN-DIAN, and 施君典. "Adaptive decision feedback equalizer for tdma mobile radio receiver." Thesis, 1989. http://ndltd.ncl.edu.tw/handle/97955454574045569248.
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Tsai, Ming-hung, and 蔡明宏. "Design of Artificial Neural Network Based Adaptive Channel Equalizer." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/99555841544867192611.
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碩士國立雲林科技大學
電機工程系碩士班
101
In this thesis, we use Chebyshev functional link neural network (CFLANN) architecture to design a channel equalizer for 4-quadrature amplitude modulation (4-QAM) transmission systems. System performance was compared with two commonly used channel equalizers, namely, linear-least-mean-square based equalizer (LIN) and multilayer perceptron (MLP). The nonlinear approximation characteristics of Chebyshev functions can expand the low spatial dimensions of original input signals to higher ones, so as to solve the linearly unclassifiable problem. In addition, because CFLANN uses functional expansions instead of hidden layers, it has much simpler structure, lower computational complexity and higher speed of convergence. Simulation results show that CFLANN indeed present higher convergence rate than LIN and MLP during training mode. Under the circumstances of highly nonlinear distortion and severe inter-symbol interference, the bit error rate (BER) performance of CFLANN equalizer is 3.8 dB and 7.7 dB better than MLP and LIN, respectively.
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Chen, Nancy Fang-Yih. "High-Performance Adaptive Decision Feedback Equalizer Designs for Ethernet Systems." 2004. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-1307200416371500.
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李智偉. "Improvements in convergence of the blind adaptive decision feedback equalizer." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/71009722495584364480.
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碩士國立海洋大學
電機工程學系
90
In the field of underwater acoustic communications, digital communications systems generally use binary phase shift keying (BPSK) and quadrature phase shift keying (QPSK) signaling. The multipath effects and Doppler frequency shift can be overcome by an adaptive trained decision feedback equalizer (DFE). Unfortunately, the trained DFE is inefficient, unless it is periodically retrained, for digital communications through a water channel. But it will reduce effective capacity. The blind adaptive decision feedback equalization (BADFE) exhibits good convergence rate and symbol-error rate (SER) as the conventional trained decision feedback equalizer (DFE) does, but it requires no training. In this thesis, a new method for BADFE is discussed. An recursive-least-square (RLS) algorithm is employed for updating the parameters of the recursive whitening filter in BADFE during the starting period, and the revised structure is called BADFE(RLS). BADFE(RLS) can speed up the convergence rate and get better tracking capabilities. This will improve the quality of communication and increase the capacity. Because of the advantages of the reliable, programmable and model of the DSP system, we complier the BADFE program based on TMS320C6711 DSK. The testing system uses a QPSK modulation with 12 kbps data rate and the carrier frequency is 36 kHz.
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Chen, Nancy Fang-Yih, and 陳方玉. "High-Performance Adaptive Decision Feedback Equalizer Designs for Ethernet Systems." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/w56wtg.
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碩士國立臺灣大學
電子工程學研究所
92
As the applications and prevalence of the world wide web (WWW) flourish over the past decade, the demand for higher bandwidth and data rate is skyrocketing. In August 2002, the IEEE 802.3ae task force finalized the 10-Gigabit Base LX4 Ethernet Standard. However, under multi-gigabit data rates, fiber is no longer an ideal transmission medium. This is especially the case for multi-mode fiber (MMF) used in 10G Base LX4 Ethernet systems, because it suffers from differential mode dispersion (DMD). However, conventional adaptive decision feedback equalizers (ADFE) cannot attain the bit error rate (BER) requirement of 10-12 in 10G Base LX4 Ethernet systems, because hard decision of slicers causes error propagation in the feedback loop. Soft threshold multi-layer adaptive decision feedback equalizer (STM-ADFE) designs are adopted to solve this problem. Our system simulation environment includes three representative channel impulses responses, trans-impedance amplifier (TIA), analog equalizer (AEQ), analog/digital converter (ADC), and STM-ADFE. Integration with the analog front end reduces the filter tap numbers needed in STM-ADFE. VLSI architectures of STM-ADFE are also presented. With low hardware overhead, STM-ADFE not only lowers the BER, but also reduces the bit resolution needed in the ADC from 8 to 6.
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Lin, Jui-Chieh, and 林叡杰. "Design and Implementation of a High-Speed Analog Adaptive Equalizer." Thesis, 2007. http://ndltd.ncl.edu.tw/handle/61194650493015278420.
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碩士國立臺灣大學
電子工程學研究所
95
Nowadays, data transmission operates at a very high speed. However as the transmission data rate becomes higher, the signal suffers from more severe frequency dependent magnitude loss due to the channel limited bandwidth. Analog adaptive equalizer has been proven of great use to compensate the non-ideality. Along with the adaptability, analog equalizer is capable of giving adequate boosting to time and temperature varying channel loss. In the Thesis, an analog filter fabricated in TSMC 0.18μm 1P6M CMOS technology is designed. MOS varactor and transistor in triode region acts as variable resistor is applied for adaptability.