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Sharma, Kripa. "Bilevel Equalizer Drivers for Large Lithium-Ion Batteries." University of Toledo / OhioLINK, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=toledo1564677943667852.
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Kim, Hyoung soo. "Design of silicon-based equalization techniques for band limited giga hertz channels." Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/33996.
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The object of this research is to develop a solution for band-limited channels. Backplane channels and GPON channels are investigated to apply an equalization technique. Different lengths of backplane channels are measured with different signal speeds to investigate the channel performance. Also a GPON system with different fiber lengths is designed and set up in a lab to measure the BER performance. The GPON system utilizes a Fabry-Perot laser for the most economical solution. After the circuits are fabricated, they are inserted into the system to measure the performance of the channels with equalizers. Both the backplane and the GPON system show successful channel improvement in measured eye diagrams and BER. To expedite the procedure and eventually build an adaptive system which could be inserted and self-optimizing, we found it essential to monitor the output of the equalizer. A novel analog way to achieve this goal is suggested. All the equalizers mentioned in this dissertation have one summing node to add up all the values from VGAs. This structure is very efficient, but in the event that there are too many VGAs, it draws too much current through the one node. This issue is dealt with by the design of two nine tap equalizers, which are compared to assess the difference in performance between the unbalanced structure and the balanced structure.
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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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Mayer, Kayol Soares. "NMCMA-SDD concurrent equalizer." Pontif?cia Universidade Cat?lica do Rio Grande do Sul, 2018. http://tede2.pucrs.br/tede2/handle/tede/7981.
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Coordena??o de Aperfei?oamento de Pessoal de N?vel Superior - CAPES
Em sistemas de comunica??o digital sem fio, o sinal transmitido pode ser afetado por v?rias fontes de distor??o, sendo as mais significantes o ru?do gaussiano branco aditivo (AWGN), multipercurso e efeito Doppler. Em ambientes onde a resposta de impulso do canal de comunica??o ? vari?vel no tempo, como em comunica??es m?veis, a distor??o por multipercurso ? din?mica. Juntamente com o efeito Doppler, o multipercurso din?mico pode at? mesmo eventualmente interromper completamente o link de comunica??o sem fio. Para minimizar este problema, equalizadores de canais autodidatas s?o usados para mitigar os efeitos de multipercurso e Doppler. Neste contexto, esta disserta??o prop?e um novo equalizador de canal autodidata (blind), baseado no algoritmo de m?dulo constante modificado n?o linear (NMCMA) e no equalizador Soft Direct Decision (SDD) em uma arquitetura concorrente. Este novo equalizador concorrente NMCMA-SDD ? comparado com o estado da arte, o equalizador concorrente com algoritmo de m?dulo constante (CMA)-SDD, operando sob os chamados "Brazil channels A-E", proposto pela Uni?o Internacional das Telecomunica??es (UIT). O equalizador proposto apresenta resultados significativos em compara??o com o estado da arte, possibilitando a opera??o em links de comunica??o sem fio entre ve?culos a?reos n?o tripulados (UAVs), ve?culos terrestres e em outros cen?rios de comunica??o din?mica.
In wireless digital communication systems, the transmitted signal may be affected by several sources of distortion, the most significant being Additive Whit Gaussian Noise (AWGN), multipath and Dopplr effect. In environmets where the impulse response of the communication channel is time variant, as in mobile communications, the multipath distortion is dynamic. Together with the Doppler effect, the dynamic multipath may even completely interrupt the wireless communication link. In order to solve this issue, blind channel equalizers are used to mitigate the multipath and Doppler effects. In this context, this dissertation proposes a novel blind channel equalizer, based on the Nonlinear Modified Constant Modulus Algorithm (NMCMA) and on the Soft Direct Desicion (SDD) equalizers in a concurrent architecture. This novel NMCMA-SDD concurrent equalizer is compared with the state of the art, the Constant Modulus Algorithm (CMA)-SDD concurrent equalizer, over the so-called "Brazil channels A-E", proposed by the International Telecommunication Union (ITU). The proposed equalizer presents significant results when compared with the state of the art, making possible its operation in wireless communication links for Unmanned Aerial Vehicles (UAVs), terrestrial vehicles, and others dynamic communication scenarios.
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Karr, Jolanda Tracie. "Environmental education: The equalizer." CSUSB ScholarWorks, 2005. https://scholarworks.lib.csusb.edu/etd-project/2860.
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Afran, Md Shah. "Frequency Domain Equalizer for Aeronautical Telemetry." International Foundation for Telemetering, 2015. http://hdl.handle.net/10150/596444.
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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, NVThis paper presents a frequency domain equalization (FDE) technique for aeronautical telemetry channels. The FDE has significantly lower computational complexity compared to its time-domain counterpart, however both are found to exhibit almost identical performance. A cyclic prefix is generally needed to implement the FDE. In this paper, we exploit the repetition of iNET preamble and ASM bits in place of cyclic prefix.
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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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Xingwen, Ding, Zhai Wantao, Chang Hongyu, and Chen Ming. "CMA BLIND EQUALIZER FOR AERONAUTICAL TELEMETRY." International Foundation for Telemetering, 2016. http://hdl.handle.net/10150/624262.
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In aeronautical telemetry, the multipath interference usually causes significant performance
degradation. As the bit rate of telemetry systems increases, the impairments of multipath
interference are more serious. The constant modulus algorithm (CMA) blind equalizer is effective
to mitigate the impairments of multipath interference. The CMA adapts the equalizer coefficients
to minimize the deviation of the signal envelope from a constant level. This paper presents the
performances of the CMA blind equalizer applied for PCM-FM, PCM-BPSK, SOQPSK-TG and
ARTM CPM in aeronautical telemetry.
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Chandramouli, Soumya. "A Novel Analog Decision-Feedback Equalizer in CMOS for Serial 10-Gb/sec Data Transmission Systems." Diss., Georgia Institute of Technology, 2007. http://hdl.handle.net/1853/19847.
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This dissertation develops an unclocked receiver analog decision-feedback equalizer (ADFE) circuit architecture and topology and implements the circuit in 0.18-um CMOS to enable 10-Gb/sec serial baseband data transmission over FR-4 backplane and optical fibre. The ADFE overcomes the first feedback-loop latency challenge of traditional digital and mixed-signal DFEs by separating data re-timing from equalization and also eliminates the need for clock-recovery prior to decision-feedback equalization.
The ADFE enables 10-Gb/sec decision-feedback equalization using a 0.18-um CMOS process, the first to do so to the author s knowledge. A tuneable current-mode-logic (CML) feedback-loop is designed to enable first post-cursor cancellation for a range of data-rates and to have external control over loop latency over variations in process, voltage and temperature. CML design techniques are used to minimize current consumption and achieve the required voltage swing for decision-feedback to take place. The all-analog equalizer consumes less power and area than comparable state-of-the art DFEs.
The ADFE is used to compensate inter-symbol interference (ISI) for 20 inches of FR-4 backplane and 300 m of multi-mode fibre at 10-Gb/sec. The ADFE also extends the reach of single-mode fibre at 10-Gb/sec to 120 km. The work described in this dissertation advances the state-of-the-art in equalization solutions for multi-Gb/sec serial data transmission and can find applications in several of the 10-Gb/sec Ethernet standards that have been approved recently. The contributions of this work toward future research are also discussed.
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Tressler, Neera. "Equalizer design for discrete wavelet multitone transceiver." Thesis, University of Ottawa (Canada), 2002. http://hdl.handle.net/10393/6182.
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Discrete Wavelet Multi-tone (DWMT) is a multi-carrier technique that uses Wavelet Transform (WT) for modulation. A DWMT communication system is designed and an attempt is made to optimize the equalizer design and the structure of the DWMT receiver. Three plausible receiver designs consisting of a time domain equalizer (TEQ), several frequency domain equalizers (FEQ) and a combination of a simple TEQ and simple FEQs, are examined. The Linear equalizer (LE) is used as the TEQ, whereas the LE and the Decision Feedback Equalizer (DFE) are used as FEQs. These designs are simulated, tested on different twisted pair Very high-speed Digital Subscriber Loop (VDSL) channels and compared with each other in terms of their complexity and performance. The results indicate that the receiver designs consisting of only TEQ or only FEQs do not give a good signal reconstruction, in spite of using a large number of taps in the equalizers. On the other hand, the receiver structure consisting of a combination of a TEQ and FEQs was found to give a better performance, with lesser complexity in both the TEQ and the FEQs. A few taps in TEQ are found to be sufficient for eliminating most of the ISI in the signal. After demodulation, an FEQ with a few taps for each subchannel is sufficient to equalize the signal further and give a much lower Symbol Error Rate SER, as compared to both the previous structures. The DFE was found to give better performance than the LE. We conclude that the receiver design consisting of a simple TEQ and simple subchannel DFE-based FEQs, is most suitable for the DWMT transceiver among the three different designs tested for a DWMT system.
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Schlagenhaufer, Ramon. "Equalizer structures for spread spectrum multiuser systems." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2001. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq64839.pdf.
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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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Mehreteab, Deborah. "Strategisk prissättningsstrategi med stöd av prismodellen Equalizer." Thesis, Uppsala universitet, Institutionen för teknikvetenskaper, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-393027.
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Pricing strategy for a company, business or organization is crucial in all industries. Being able to offer attractive prices affects suppliers and customers. Understanding a good and flexible pricing is essential for a company to get good results. Pricing models are a concept used to deepen the analysis of how pricing strategically affects organizations. In SaaS and startups, pricing models and pricing strategy are not always the focus, a common misconception companies do is focusing more on the product and therefore ending up in a market-based pricing strategy automatically because it is the easiest way of pricing. This is a problem because the product might be of higher value that a customer is willing to pay more for. The purpose of this study is to deliver a pricing strategy for the company Pickit. The study was divided into two sub-areas; customer and competitor. The project has been delimited by not examining which pricing models or pricing strategies competitors use. The work has also been limited to not following up on whether the solution will be accepted or implemented in the business. The study began with interviews with the company to create a deeper understanding of the problem. Subsequently, various interviews were conducted with customers and employees, literature studies, data collection of potential customers and competitors. Observations in the office were also made in order to study the culture. From the customer analysis, it was found that three out of forty customers’ accounts for 44% of total sales. From the competitor survey it was found that the competitors today are fulfilling the customer need in a broader way than Pickit can offer, but Pickit´s product has a value that is unique on the market. The project's conclusion suggests Pickit to base their pricing from a customerbased perspective with elements of competitor-based pricing in order to stay relevant on the market. The Equalizer pricing model supports the chosen pricing by visually making it clear to see what is included in Pickits pricing.Prissättningsstrategi för ett företag, verksamhet eller organisation är avgörande i alla branscher och industrier. Att kunna erbjuda lockande priser påverkar både leverantörer och kunder vilket kan avgöra affärer. Att ha en förståelse för en bra och flexibel prissättning är väsentligt för ett företag ska få bra resultat. Prismodeller är ett koncept för att fördjupa analysen av hur prissättning strategiskt påverkar organisationer. Inom SaaS och startup är prismodeller och prisstrategi inte alltid i mest fokus, en vanlig miss är att företagen fokuserar mer på produkten och hamnar i en marknadsbaserad prissättningsstrategi automatiskt eftersom prissättning inte anses som det viktigaste, vilket gör att företagen tar den enkla lösningen först. Syftet med denna studie är att leverera en prissättningsstrategi för företaget Pickit. Studien delades in i två delområden; kund och konkurrent. Projektet har avgränsats genom att inte undersöka vilka prismodeller eller prissättningsstrategier konkurrenterna använder. Arbetet har även avgränsats till att inte följa upp huruvida lösningen kommer att accepteras eller implementeras i verksamheten. Studien inledes med intervjuer med företaget för att skapa en djupare förståelse för problemet. Därefter gjordes diverse intervjuer med kunder och anställda, observationer på kontoret för att exempelvis studera deras inställning mot förändring, datainsamling av potentiella kunder samt konkurrenter. Av kundanalysen kom det fram att tre av fyrtio kunder står för 44% av den totala omsättningen. Av konkurrentkartläggningen kom det fram att kundbehovet som konkurrenterna uppfyller i dagsläget bredare än vad Pickit kan erbjuda men att det värdeerbjudande Pickit erbjuder på marknaden är unikt. Projektets slutsats innebar att prissättningsstrategin som Pickit borde utgå från är kundbaseradprissättning med inslag av konkurrentbaserad prissättning för att hålla sig relevant på marknaden. Prismodellen Equalizer stödjer arbetet genom att visuellt göra det tydligt att se vad Pickit tar betalt och stärkte slutsatsen om vilken strategi som passar företaget.
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Ryan, Brooks C. "Are Schools the Great (Noncognitive Skills) Equalizer?" The Ohio State University, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=osu1305920898.
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Larijani, P. Roohi (Parsya Roohi) Carleton University Dissertation Engineering Systems and Computer. "Radial basis function equalizer for mobile channels." Ottawa, 1994.
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Brown, Tim. "PIC controlled two-band stereo audio equalizer." Click here to view, 2009. http://digitalcommons.calpoly.edu/eesp/14/.
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Thesis (B.S.)--California Polytechnic State University, 2009.Project advisor: Dennis Derickson. Title from PDF title page; viewed on Feb. 4, 2010. Includes bibliographical references. Also available on microfiche.
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Feng, Xue Ph D. Massachusetts Institute of Technology. "Sparse equalizer filter design for multi-path channels." Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1721.1/75657.
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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 81-82).
In this thesis, sparse Finite Impulse Response (FIR) equalizers are designed for sparse multi-path channels under a pre-defined Mean Squared Error (MSE) constraint. We start by examining the intrinsic sparsity of the Zero Forcing equalizers and the FIR Minimum MSE (MMSE) equalizers. Next the equalization MSE is formulated as a quadratic function of the equalizer coefficients. Both the Linear Equalizer (LE) and the Decision Feedback Equalizer (DFE) are analyzed. Utilizing the quadratic form, designing a sparse equalizer under a single MSE constraint becomes an 10-norm minimization problem under a quadratic constraint, as described in [2]. Three previously developed methods for solving this problem are applied, namely the successive thinning algorithm, the branch-and-bound algorithm, and the simple linear programming algorithm. Simulations under various channel specifications, equalizer specifications and algorithm specifications are conducted to show the dependency of the sparsity on these factors. The channels include the ideal discrete multipath channels and the Vehicular A multi-path channels in both the Single-Input-Single- Output (SISO) and the Multiple-Input-Multiple-Output scenarios. Additionally, the sparse FIR equalizer is designed for MIMO channels under two MSE constraints. This is formulated as an 10-norm minimization problem under two quadratic constraints. A sub-optimal solution by decoupling the two constraints is proposed.
by Xue Feng.
S.M.
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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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Gonzalez, Fitch David E. "A novel OFDM Blind Equalizer: Analysis and Implementation." Thesis, Virginia Tech, 2012. http://hdl.handle.net/10919/34833.
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Link adaptation is important to guarantee robust and reliable wireless communications with-
out wasting valuable radio resources. This technique has become more feasible with the
recent appearance of Software Defined Radios (SDRs), which allow easy reconfiguration of
their parameters via software. As the environment changes over time, the transmitter needs
to be able to effectively estimate its performance under different radio input parameters to
be able to find a close to optimal solution. In most wireless communications, an equalizer
is implemented at the receiver to estimate the channel impulse response. This estimate can
be fed back to the transmitter via a feedback channel, which can in turn help generate a
sub-optimal transmission solution for the current situation.
In this thesis, a link adaptation method is proposed that uses Orthogonal Frequency-Division
Multiplexing (OFDM) in conjunction with blind channel estimation. With the use of OFDM,
it can be assumed that the frequency fading at each subcarrier is approximately flat. In
addition, under the assumption that the channel is quasi-stationary, the Bit Error Rate
(BER) at each subcarrier can be estimated by using the well-known BER formulas for an
Additive White Gaussian Noise (AWGN) channel. However, the effect of imperfect channel
estimation must also be taken into account.
A novel OFDM blind channel estimator is developed. Finally, both simulations and real
over-the-air results are presented.Master of Science
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Kwan, Man-Wai. "Minimal transmit redundancy FIR precoder-equalizer systems design /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20KWAN.
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Sigfridsson, Simon. "Is it possible to create an indistinguishable or equal frequency response between a digital equalizer and an analog emulating equalizer plug-in?" Thesis, Luleå tekniska universitet, Medier ljudteknik och upplevelseproduktion och teater, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-69129.
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This research examines if an indistinguishable, or equal, frequency response of a software plug-in that is emulating an analog equalizer can be reconstructed using a more standard digital equalizer such as one incorporated with a digital audio workstation (DAW). It is narrowed down to solely involve high frequency bands by analyzing the emulating plug-ins hi-shelving filters. A two-comparison forced choice ABX-test was conducted to verify the hypotheses and the results show that the difference between the original and the reconstructed hi-shelving filter was inaudible to the listening test participants. Further research and application for these findings is discussed
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Yu, Yong. "Time domain equalizer design based on multi-rate technique." Thesis, University of Ottawa (Canada), 2002. http://hdl.handle.net/10393/6310.
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Discrete Multitone is adopted in Digital Subscriber Line to offer high-speed data communication on the UTP channel. To combat the channel distortion, DMT system employed Time Domain Equalizer and Cyclic Prefix. In this thesis, we analyze different existing time domain equalizer design methods and their performances are compared based on our simulation results. Modification of the current method is proposed and significant performance improvement is obtained. Multi-rate equalization is studied theoretically, which enables us to achieve zero ISI channels that can be used not only in DMT system but also in all other communication systems. Our simulation results show that the new design method is superior and practical.
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Song, Zhonghe. "VHDL design and implementation of high speed blind equalizer." Thesis, University of Ottawa (Canada), 2004. http://hdl.handle.net/10393/26775.
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Blind equalization has significant advantages over conventional adaptive equalization because of its self-recovery ability. This ability makes blind equalization the best choice in broadcasting type communication. With increased demands for broadband data communication, the speed of blind equalization is continuously increasing. The main challenge lies in designing a high speed blind equalizer using current VLSI technology.
Different blind equalization algorithms were investigated for this thesis. Comparisons of their performances and hardware complexities were performed. The CM algorithm offers excellent performance and the most robust property. A pipelined transposed direct form structure equalizer based on the CM blind equalization algorithm was proposed and designed in a VHDL model. Its performance was verified and the finite wordlength effect was investigated. VHDL model was synthesized using 0.25 mum technology.
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Afran, Md Shah, Mohammad Saquib, and Michael Rice. "SPARSE MMSE EQUALIZER FOR GTR-STBC IN AERONAUTICAL TELEMETRY." International Foundation for Telemetering, 2017. http://hdl.handle.net/10150/626962.
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This paper investigates the performance of sparse minimum mean squared error (MMSE) equalizer for generalized time-reversed space-time block codes (GTR-STBC) in aeronautical telemetry. GTR-STBC equipped with MMSE equalizer performs the best trade-off between the signal-tonoise ratio and inter-symbol interference by allocating unequal power over aeronautical telemetry channels. However, aeronautical telemetry channels are in general consists of larger delay spreads which make the MMSE equalization of aeronautical channels with GTR-STBC computationally complex. Interestingly enough, in spite of larger delays aeronautical channels are made of few sparsely distributed multipaths and therefore their MMSE equalizers are highly compressible. In this paper, compressed sensing based greedy algorithm is used for the design of sparse MMSE equalizer and a convex curve-fitting algorithm is used to find the sub-optimum power allocation parameter at the same sparsity level for GTR-STBC. Our simulation results show that 75-90% of the non-zero equalizer taps can be reduced with a slight relaxation of the mean-squared error (or equivalentlyslight degradationof bit-errorrate performance). It isalso observedthat the optimum transmitter power profile for the sparse MMSE equalizer is different than that of the non-sparse equalizer.
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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.
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碩士國立臺灣大學
電子工程學研究所
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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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YOU, YUE-E., and 游月娥. "Group-delay equalizer." Thesis, 1986. http://ndltd.ncl.edu.tw/handle/65570739374198491685.
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Paul-Yuan, Chen, and 陳柏淵. "Blind Equalizer Research." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/03611258805362184207.
Full textAbstract:
碩士中原大學
電子工程研究所
86
Blind equalizers have more been attractive in recent years. Itsapplications, such as mobile communication and HDTV, need the char-acteristics of blind equalizers of fast rate of convergence. But thetraditional blind equalizers have the problem of slow rate of conver-gence. The objective of this paper is to design a blind equalizerthat has the char-acteristics of fast rate of convergence. Traditional blind equalization methods are to design a new cri-teria, different from the mean-square error (MSE) criteria used fortrained equalizers, and apply a gradient-search algorithm to optimizethe selected criterion. The drawback of the class of the equalizers is(1) converging to a local minima, and (2) slow rate of convergence.In a more difficult en-vironment, such as mobile digital commu-nications, the class of the blind equalizers cannot converge in sometime. Accordingly the class of the blind equalizers cannot used in suchenvironments. Blind equalization methods mostly use the characteristics ofcyclostationary signals in recent years because using such char-acteristics can overcome the problem of slow rate of convergenceso as to converge fast. In this paper, we use the cyclic Wiener filteringmethod which is not only converge fast, but also simple in architecture. Split-path LMS filter structure can improve the convergence rate, butapplying directly such filter structure to blind equalizers cannotachieve expected effect. Then we modify the structure so as to speed upthe convergence rate of blind equalizers. We use the cyclic Wienerfiltering method and replace the filter part of the blind equalizerswith the split-path adaptive filter structure so as to achieve fasterconvergence rate. Simulation results demonstrate the convergencerate of the split-path blind equalizers really faster than the traditionalblind equalizers.
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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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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.
Full textAbstract:
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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Chen, Yu-Sheng, and 陳育聖. "Reduced-Complexity Equalizer Design." Thesis, 2000. http://ndltd.ncl.edu.tw/handle/21907212384809887119.
Full textAbstract:
碩士國立雲林科技大學
電子工程與資訊工程技術研究所
88
This thesis aims at the development of various equalization methods for efficient VLSI realization, including 1. Time-domain equalization for asymmetric digital subscriber line (ADSL), 2. Reduced-complexity equalization via convolution decomposition (CD), and block processing, 3. Blind equalization based on LMS (least mean square error) and constant modulus (CM) criteria. We work out a new blind equalization method for acceleration of convergence based on the signed Godard algorithm (SGA) and decision feedback. The averaged stochastic gradient technique is also studied extensively to sustain a low-complexity design with improved convergence performance of the LMS method. For equalization in ADSL, we employ bisection search to select discrete step sizes adaptively for low-complexity VLSI implementation.
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Xue, Mei-Zhen, and 薛美珍. "CPLD Based Battery Equalizer." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/z6x8mb.
Full textAbstract:
碩士國立高雄海洋科技大學
微電子工程研究所
100
In this thesis, a battery charge and management system applied for Li-Fe battery set is developed. The developed battery management system consists of an active battery equalizer and a forward type buck charger. Since the high charging/discharging current characteristic of Li-Fe batteries is superior to that of the lead-acid batteries, Li-Fe batteries have gradually been applied to the applications of power battery. Because the voltage of a single Li-Fe battery is low, several Li-Fe batteries must be connected in series and form a battery set to supply enough voltage. However, characteristic of individual battery is unable to be uniform, the voltage of individual battery in the battery set will be different each other under the charge/discharge process. It will reduce the cycle life and the capacity of the battery set. The active battery equalizer is developed for balancing the individual battery voltage of the Li-Fe battery set in the charging/discharging process. The developed active battery equalizer is composed of eight power electronic switches and an inductor. The inductor is energized by the battery set and then charge the battery whose voltage is lowest by properly controlling the power electronic switches. In this way, balancing the voltage of batteries in the battery string is performed by the developed active battery equalizer. A Complex Programmable Logic Device (CPLD) based prototype is developed to verify the performance of proposed active battery equalizer. Besides, performance and life time of a battery set is also dependent on the battery charger. In this thesis, the forword type buck charger is developed for charging the Li-Fe battery set. It can generate an adjustable output DC voltage to properly charge the Li-Fe battery. In addition, the input current of the forword type buck charger is near sinusoidal and in phase with the utility voltage so as to perform unit power factor. A digital signal processor (DSP) based prototype is developed to verify the performance of proposed forword type buck charger.
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"An Efficient DOCSIS Upstream Equalizer." Thesis, 2014. http://hdl.handle.net/10388/ETD-2014-03-1454.
Full textAbstract:
The advancement in the CATV industry has been remarkable. In the beginning, CATV provided a few television channels. Now it provides a variety of advanced services such as video on demand (VOD), Internet access, Pay-Per-View on demand
and interactive TV. These advances have increased the popularity of CATV manyfold. Current improvements focus on interactive services with high quality. These
interactive services require more upstream (transmission from customer premises to cable operator premises) channel bandwidth.
The flow of data through the CATV network in both the upstream and downstream directions is governed by a standard referred to as the Data Over Cable Service Interface Specification (DOCSIS) standard. The latest version is DOCSIS 3.1,
which was released in January 2014. The previous version, DOCSIS 3.0, was released in 2006.
One component of the upstream communication link is the QAM demodulator. An important component in the QAM demodulator is the equalizer, whose purpose
is to remove distortion caused by the imperfect upstream channel as well as the residual timing offset and frequency offset. Most of the timing and frequency offset are corrected by timing and frequency recovery circuits; what remains is referred to
as offset.
A DOCSIS receiver, and hence the equalizer within, can be implemented with ASIC or FPGA technology. Implementing an equalizer in an ASIC has a large nonrecurring
engineering cost, but relatively small per chip production cost. Implementing equalizer in an FPGA has very low non-recurring cost, but a relatively high per chip cost. If the choice technology was based on cost, one would think it would depends only on the volume, but in practice that is not the case. The dominant factor when it comes to profit, is the time-to-market, which makes FPGA technology the
only choice.
The goal of this thesis is to design a cost optimized equalizer for DOCSIS upstream demodulator and implement in an FPGA. With this in mind, an important objective is to establish a relationship between the equalizer’s critical parameters
and its performance. The parameter-performance relationship that has been established in this study revealed that equalizer step size and length parameters should be
1/64 and approximately 20 to yield a near optimum equalizer when considering the MER-convergence time trade-off.
In the pursuit of the objective another relationship was established that is useful in determining the accuracy of the timing recovery circuit. That relationship establishes the sensitivity both of the MER and convergence time to timing offset.
The equalizer algorithm was implemented in a cost effective manner using DSP Builder. The effort to minimize cost was focused on minimizing the number of multipliers.
It is shown that the equalizer can be constructed with 8 multipliers when the proposed time sharing algorithm is implemented.
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Yu, Ya-Ju, and 游雅如. "Equalizer Design for Wireless Communications." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/75238912281450211895.
Full textAbstract:
碩士國立交通大學
電子工程系
89
The performance of adaptive equalizers using the normalized least mean square algorithm (NLMS) with QPSK modulation technique for high data rate wireless communications is investigated. Multi-ray tapped delay line Rayleigh channel model is used for propagation analysis. The tapped delay and average power of the each path refer to ITU tapped delay line indoor channel model. The Jakes’ model is used to generate the Rayleigh distribution of the each path. The effect of using linear transversal equalizer (LTE) and decision feedback equalizer (DFE) in complex structure for such channel model is evaluated and compared with that without equalizer. The influence of some important parameters, such as tap number of the equalizers, and step size setting of the adaptive algorithm, is examined. The convergence speed and final convergence variance of the NLMS with different step size for LTE and DFE is presented. Design specifications can then be adjusted according to system requirement. A modified NLMS is proposed and had been implemented to reduce hardware computational complexity.
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Chen, Bo-jheng, and 陳博正. "Design of an Intelligent Equalizer." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/90618989422930725293.
Full textAbstract:
碩士國立臺灣科技大學
電機工程系
98
This research is on the basis of psychoacoustic theory. According to Fletcher-Munson’s Equal Loudness Curve, ears to the sound are characterized by nonlinearity, and same loudness in frequency requires different sound press level, which highlights the concept that ears have balanced feeling to different loudness in sound. Consequently, the study intends to set up a table that presents the compensative relationship between volume and equalizer by testing users’ actual sound in examining the volume variation of pure tone signal. Based on the table using C language to implemented, the study shows that equalizer makes adjustments automatically in compensating different volume while users adjust volume.
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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.
Full textAbstract:
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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Lan, Yi-Yang, and 藍義陽. "Volterra-based Decision Feedback Equalizer." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/77614132011693728845.
Full textAbstract:
碩士國立交通大學
電信研究所
81
The advantages of a conventional decision feedback equa- lizer(CDFE) is its simplicity in design and in implementa- tion. However, the feedback part of CDFE is a linear equa- lizer and, hence,the decision regions in the signal space are delimited by hyperplanes. This property limits the performance of the system. In the articles, we propose a new adaptive nonlinear equalizer, called Volterra-based decision feedback equalizer, which is capable of forming nonlinear decision boundaries in the signal space. As a result, the performance limit in the CDFE can effectively be overcome. Especially, it has outstanding performance whenever the transmission channel is nonlinear. Moreover, conventional adaptation algorithms can be directly applied to the new structure. The simulation result demonstrate that the new structure can be an effective alternative in contrast to the CDFE.
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Sun, Jie. "Equalizer design for wireless communications." 2010. https://scholarworks.umass.edu/dissertations/AAI3427572.
Full textAbstract:
As the speed of communication systems increases into the gigabits per second range, new applications such as high definition video streaming and real time imaging systems become feasible in an indoor environment. Wireless transfer rates for these applications are far in excess of what can be accommodated in the currently used bands at 2.4 GHz and 5.2 GHz. An obvious solution is to resort to the newly opened millimeter wave bands. Unfortunately 60 GHz systems exhibit many challenges that have made them difficult to deploy [8]. One of these challenges is the severe frequency selective fading due to multiple path reflections. To recover the transmitted signal from the effects of fading, a carefully designed channel equalization scheme must be deployed at the receiver. For systems with multi-gigabit per second transmission rates, the heating of the components is a crucial factor for circuit design. Because the analog circuits have higher power efficiency than digital signal processing (DSP), people are looking for solutions to use continuous time signal processing technology for very broadband signals. In the first part of this dissertation, we propose two “semi-analog” channel equalizer designs for communications at multi-gigabit per second data rate. The first equalizer is designed from frequency domain. We propose a stable equalizer whose Laplace transform is close to the ideal equalizer except for some constant group delay. The second equalizer is designed from time domain analysis. It targets at the multipath channel and recovers transmitted signal by removing all the reflected ones from the received signal. In the second part, a continuous time Kalman filter-based simple smoother is discussed to recover the transmitted data sequences. Although the continuous time Kalman filter is designed for Additive white Gaussian noise only, we propose a new dynamic system under this environment which has the same performance as the matched filter. This provides more flexibility for system design. In the latter part of the dissertation, we propose a novel blind equalizer design. Adaptive equalization generally requires an initial training period during which a known data sequence is transmitted. However in wireless communications, channel changes very frequently, which makes it impractical or impossible to send a training sequence. Blind equalization is an essential technique to remove the intersymbol interference introduced by the channel when training sequence is unavailable. Our approach is quite different from traditional stochastic gradient descent algorithm. We use eye diagram technique and Viterbi decoding algorithm to recover the transmitted data sequences without knowing the data transmission rate.
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許哲嘉. "FIR Equalizer Design Using Convex Optimization." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/51022378848924851578.
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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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Su, Yu-Chang, and 蘇裕彰. "Equalizer Design for HiperMAN Wireless Network." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/96529245492196152973.
Full textAbstract:
碩士國立中興大學
電機工程學系所
94
HiperMAN (High Performance Radio Metropolitan Area Network) is a standard created by ETSI (European Telecommunications Standards Institute) BRAN (Broadband Radio Access Networks) group to provide inter-operable broadband fixed wireless communication access in the 2 - 11GHz radio frequency bands across Europe HiperMAN is optimised for packet switched networks, and fixed and nomadic applications, primarily in the residential and small business user environments. In HiperMAN system, it is composed of three parts, including channal coding, modulation and orthogonal frequency division multiplexing (OFDM) system. The objective of this project is model and simulate the ETSI HiperMAN OFDM physical layer using MATLAB. The simulation will compare the performance of two different receiver models. In the first method, we only use the pilot signal to adjust the parameters of the equalizer, and in next we use all data passed by channel to adjust the parameters.
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Chuang, Yuan-Shin, and 莊源欣. "Design of 802.11a Receiver and Equalizer." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/78947459551708754685.
Full textAbstract:
碩士國立交通大學
電子工程系
91
In the thesis an effective algorithm for equalization, carrier frequency offset compensation, and sampling clock offset compensation for OFDM based wireless LAN specified in 802.11a are proposed. In the design of equalization, two long preamble specified in front of one packet is used to estimate channel response. A new method is proposed to estimate channel response more accurately. Two long preambles and pilot signals are used to estimate the carrier frequency offset and sampling clock offset compensation. With the SNR input equal to 22dB, the residual amount of CFO, SCO is less than 0.1ppm and 1ppm. FPGA is used to implement our design.
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Wang, Jui-Chiang, and 王瑞強. "HDTV EQUALIZER CHIP DESIGN AND IMPLEMENTATION." Thesis, 1997. http://ndltd.ncl.edu.tw/handle/96774609140338662758.
Full textAbstract:
碩士大同工學院
電機工程學系
85
Signal distortions can come from the transmission channel or from imperfect components within the receiver. So channel equalization reduces bit detection error in the receiver by compensation for such nonideal channel characteristics. In this thesis, we realize a chip of 64-tap equalizer for Grand Alliance (GA) HDTV system. We use the transversal structure, Mean-Squared Error (MSE) criterion, and Least-Mean-Square (LMS) algorithm to design this equalizer. Finally, the equalizer is implemented and designed by CADENCE tool with cell library, COMPASS06. The working frequency of the circuit is 13MHz. The hierarchical layout area of this chip is 11335.1um×11245.4 um.
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Wu, Kei-Lin, and 鄔侃陵. "DOWNLINK EQUALIZER FOR 3G WCDMA SYSTEM." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/98964754040189069591.
Full textAbstract:
碩士國立臺灣大學
電信工程學研究所
90
In 3G WCDMA downlink, the effect of the wireless communication multi-path channel will destroy the orthogonality of different user’s spreading code, thus the multiple-access interference (MAI) arises and degrades the system performance. In demand of high data rates in 3G cellular system, how to efficiently reduce the MAI to improve the system performance is the most important thing. In this thesis, we propose using a chip-level equalizer to oppose the multi-path channel and thus the orthogonality of different user can be recovered. Furthermore, we suggest using the diversity technique to help the equalizer perform much better. The most important factor which will affect an equalizer’s performance is how to adjust the equalizer’s coefficients according to various communication channel. In this part, we first derive a matrix solution of equalizer’s coefficients. This matrix solution needs to know the information of communication channel. Thus, we introduce a least-square channel estimator to estimate the communication channel. Although the chip-level equalizer with the matrix solution to calculate the equalizer’s coefficients can efficiently reduce the MAI, it will need to calculate the inversion of a large rank matrix. This is a big problem in implementing this chip-level equalizer on hardware. Thus, we establish two adaptive algorithms to calculate the equalizer’s coefficients based on the structure of rank-reduced multi-stage Wiener filter. From the implement of these adaptive algorithms, we can obtain the balance between the system performance and computational complexity.
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Ching-Her, Huang. "2nd / 3rd Generation Mobile Phone Equalizer Study." 2007. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-1601200701505400.
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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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GUO, FANG-MING, and 郭芳名. "Bridged tap equalizer for digital subscriber loop." Thesis, 1987. http://ndltd.ncl.edu.tw/handle/24332667781555131156.
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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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Ma, Ying-Hao, and 馬英豪. "Channel equalizer for DVB-T/H System." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/86554567512524939140.
Full textAbstract:
碩士國立交通大學
電子工程系所
94
In this thesis, we introduce the channel estimation algorithm for DVB-T/H system, and COFDM basedband receiver chip for DVB-T/H applications. This chip is implemented with 0.18μm cell library and tapped out in Jun. 2005. The architecture is established according to the standard and several channel impairments. We propose the adaptive channel estimator for pilot signal which can average out the noise effects under portable environments. Furthermore, the channel response is estimated by means of two-dimension interpolation of scattered pilots. We analyze several polynomial interpolation methods under channels specified by standards. In architecture part, we can find the complex division is dominant the equalizer in cost and power. So we proposed the improved architecture for simplified the division architecture which hardware can be saved by 90.5% and power saved by 59.9% in divider itself.
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Gao, Zai-Sheng, and 高再生. "A New Bayesian Equalizer Using Polynomial Filters." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/80384174350540480315.
Full textAbstract:
碩士國立交通大學
電機與控制工程系
87
In this thesis, we propose a new Bayesian equalizer using polynomial filters. Conventional Bayesian equalizers are designed based on the linear channel, however, the channel is nonlinear in practice. Hence, to improve the performance of the equalizer, we propose a new equalizer which combines the Bayesian equalizer and the polynomial filter. We also use the polynomial filter to model the magnetic recording channel and the proposed equalizer is empolyed for the detection of the read signal from this magnetic recording channel with the effect of partial earsure. Simulation results show that the new Bayesian equalizer does outperform the conventional Bayesian equalizer.