Dissertations / Theses: 'Equalization'

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Díguele, Daniel. "Blind equalization /." Online version of thesis, 1994. http://hdl.handle.net/1850/11701.

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Olasz, Elizabeth Barbara. "Blind phase equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp05/nq20763.pdf.

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3

Konuskan, Cagatay. "Turbo Equalization for HSPA." Thesis, Linköping University, Department of Electrical Engineering, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-54640.

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New high quality mobile telecommunication services are offered everyday and the demand for higher data rates is continuously increasing. To maximize the uplink throughput in HSPA when transmission is propagated through a dispersive channel causing self-interference, equalizers are used. One interesting solution, where the equalizer and decoder exchange information in an iterative way, for improving the equalizer performance is Turbo equalization.

In this thesis a literature survey has been performed on Turbo equalization methods and a chosen method has been implemented for the uplink HSPA standard to evaluate the performance in heavily dispersive channels. The selected algorithm has been adapted for multiple receiving antennas, oversampled processing and HARQ retransmissions. The results derived from the computer based link simulations show that the implemented algorithm provide a gain of approximately 0.5 dB when performing up to 7 Turbo equalization iterations. Gains up to 1 dB have been obtained by disabling power control, not using retransmission combining and utilizing a single receiver antenna. The algorithm has also been evaluated considering alternative dispersive channels, Log-MAP decoding, different code rates, number of Turbo equalization iterations and number of Turbo decoding iterations.

The simulation results do not motivate a real implementation of the chosen algorithm considering the increased computational complexity and small gain achieved in a full featured receiver system. Further studies are needed before concluding the HSPA uplink Turbo equalization approach.

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Hooli, K. (Kari). "Equalization in WCDMA Terminals." Doctoral thesis, University of Oulu, 2003. http://urn.fi/urn:isbn:9514271831.

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Abstract Conventional versions of linear multiuser detectors (MUD) are not feasible in the wideband code division multiple access (WCDMA) downlink due to the use of long scrambling sequences. As an alternative, linear channel equalizers restore the orthogonality of the spreading sequences lost in frequency-selective channels, thus, suppressing multiple access interference (MAI) in the WCDMA downlink. In this thesis, linear channel equalizers in WCDMA terminals are studied. The purpose of the thesis is to develop novel receivers that provide performance enhancement over conventional rake receivers with an acceptable increase in complexity, and to validate their performance under WCDMA downlink conditions. Although the WCDMA standard is emphasized as the candidate system, the receivers presented are suitable for any synchronous direct sequence code division multiple access downlink employing coherent data detection and orthogonal user or channel separation. Two adaptive channel equalizers are developed based on the constrained minimum output energy (MOE) criterion and sample matrix inversion method. An existing equalizer based on the matrix inversion lemma is also developed further to become a prefilter-rake equalizer. Performance analysis is carried out for equalizers trained using a common pilot channel and for the channel response constrained MOE (CR-MOE) and sample matrix inversion (SMI) based equalizers developed in the thesis. The linear minimum mean square error (LMMSE) channel equalizer, which assumes a random scrambling sequence, is shown to approximate the performance of the LMMSE MUD. The adaptive CR-MOE, SMI-based, and prefilter-rake equalizers are observed to attain performance close to that of an approximate LMMSE channel equalizer. The equalizers considered are also shown to be suitable for implementation with fixed-point arithmetic. The SMI-based equalizer is shown to provide good performance and to require an acceptable increase in complexity. It is also well suited for symbol rate equalization after despreading, which allows for computationally efficient receiver designs for low data rate terminals. Hence, the SMI-based equalizer is a suitable receiver candidate for both high and low data rate terminals. Adaptive equalizers are considered in conjunction with forward error correction (FEC) coding, soft handover, transmit diversity and high speed downlink packet access (HSDPA). The adaptive equalizers are shown to provide significant performance gains over the rake receiver in frequency selective channels. The performance gains provided by one antenna equalizers are noted to decrease near the edges of a cell, whereas the equalizers with two receive antennas achieve significant performance improvements also with soft handover. The performance gains of one or two antenna equalizers are shown to be marginal in conjunction with transmit antenna diversity. Otherwise the equalizers are observed to attain good signal-to-noise-plus-interference ratio performance. Therefore, they are also suitable receiver candidates for HSDPA.

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Lee, Kah Ping. "Turbo equalization in wireless communication." Diss., Columbia, Mo. : University of Missouri-Columbia, 2005. http://hdl.handle.net/10355/5844.

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Thesis (M.S.)--University of Missouri-Columbia, 2005.
The entire dissertation/thesis text is included in the research.pdf file; the official abstract appears in the short.pdf file (which also appears in the research.pdf); a non-technical general description, or public abstract, appears in the public.pdf file. Title from title screen of research.pdf file viewed on (July 11, 2006) Includes bibliographical references.

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Ringholm, Magnus. "Electronegativity equalization in molecular mechanics." Thesis, Norges teknisk-naturvitenskapelige universitet, Institutt for kjemi, 2009. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-6860.

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7

Yang, Jian. "Multimodulus algorithms for blind equalization." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq25191.pdf.

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8

Lopes, Renato da Rocha. "Iterative estimation, equalization and decoding." Diss., Georgia Institute of Technology, 2003. http://hdl.handle.net/1853/15026.

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Rice, Michael, Gayatri Narumanchi, and Mohammad Saquib. "Decision Feedback Equalization for SOQPSK." International Foundation for Telemetering, 2012. http://hdl.handle.net/10150/581839.

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This paper investigates a fractionally-spaced decision-feedback equalization technique for Shaped Offset Quadrature Phase Shift Keying (SOQPSK). The kernel of the block-based feedback algorithm is to estimate the intersymbol interference and cancel it from the samples used to make the bit decisions. This process refines the bit estimates sequentially, thereby increasing the probability of obtaining accurate estimates. The simulated bit error rate performance of the decision-feedback technique shows a 1 dB improvement over MMSE-equalized SOQPSK-TG over channels derived from multipath channel measurements at Cairns Army Airfield, Ft. Rucker, Alabama and Edwards AFB, California.

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Chow, N. H. "Reduced-complexity equalization for EDGE." Thesis, University of Surrey, 2003. http://epubs.surrey.ac.uk/843777/.

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Enhanced Data Rates for GSM Evolution (EDGE) is currently being standardized as the evolution path for GSM. EDGE improves the spectral efficiency by employing an 8PSK modulation scheme with 3pi/8 rotation between symbols, which triples the GSM data rate. A Linearized-GMSK pulse shaping filter is employed to remain within the 200kHz bandwidth of GSM. In order to facilitate the ease of transition from GSM to EDGE, system parameters such as symbol rate and time slot structure remain unchanged. As a result, a network capable of EDGE can be deployed with limited investment and within a short time frame, with just an upgrade in the transceiver and the system software. The introduction of EDGE modulation has a significant effect in the receiver. The LGMSK filter introduces Inter-symbol interference whose effect becomes severe due to multi-path fading and Doppler Spreading. In addition, 8-PSK has a smaller Euclidean distance between symbols than GMSK, which makes EDGE more prone to errors. Therefore a robust equalizer is required. The research objective is to mitigate the effects of fast time-varying frequency selective fading channels in the presence of noise and interference, by optimizing the trade-off between complexity and performance. This leads to four main areas of study: Reduced-state Equalization, Pre-filtering, Reduced-state Soft Output Equalization and Joint Pre-filter, Channel and Reduced State Soft Output Data Estimation. The optimum scheme. Maximum Likelihood Sequence Estimation, based on the Viterbi Algorithm, for a 6-tap channel requires 32768 (85) trellis states. Using the techniques developed in this thesis, an implementation margin of 5.9 dB over the EDGE standard requirement is achieved with only a 2 trellis state equalizer. Subsequently, based on this low complexity structure, a new method is developed involving two stages of equalizers in cascade. With reduced decision errors and improved noise variance estimation, the two stage scheme leads to a performance surpassing the single stage, with good resistance to interference. Finally, a joint scheme of moderate complexity is developed to support the scenario of a high speed train.

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Morton, John M. "Adaptive Equalization for Indoor Channels." Thesis, Virginia Tech, 1998. http://hdl.handle.net/10919/36963.

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This thesis describes the use of adaptive equalization techniques to compensate for the intersymbol interference (ISI) that results when digital data is transmitted over a multipath radio channel. The equalization structures covered in this work are the linear transversal equalizer (LTE), the fractionally spaced equalizer (FSE), the decision-feedback equalizer (DFE), and the maximum-likelihood sequence estimation (MLSE) equalizer. This work also covers adaptive algorithms for equalization including both the least mean squares (LMS) and the recursive least squares (RLS) algorithm. All these equalizer structures and algorithms will be modeled using various simulation modules. Equalization for both stationary and mobile radio channels is considered. Stationary channels are modeled with a simple exponentially decaying profile. The mobile radio channel is represented using a two-ray Rayleigh fading model for an outdoor environment. The SIRCIM channel modeling tool is used to create channel profiles for an indoor mobile radio channel. Adaptive arrays and their similarities to linear equalizers are also studied in this thesis. The properties and performance of simple adaptive array systems using the LMS and RLS algorithms are examined through simulation. This thesis concludes with an in-depth study of the use of adaptive equalization for high-speed data systems operating in an indoor environment. Both stationary and slowly varying radio channels are examined. Simulations of DFE and MLSE equalizers operating in such a system show that both equalizer structures provide better BER performance over a system with no equalization. These simulation results also show that the MLSE equalizer provides better performance than the DFE in almost all cases, but requires a great deal more computations.
Master of Science

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Arslan, Güner. "Equalization for discrete multitone transceivers /." Digital version accessible at:, 2000. http://wwwlib.umi.com/cr/utexas/main.

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Lu, Biao. "Wireline channel estimation and equalization /." Full text (PDF) from UMI/Dissertation Abstracts International, 2000. http://wwwlib.umi.com/cr/utexas/fullcit?p3004324.

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14

Otnes, Roald. "Improved Receivers for Digital High Frequency Communications : Iterative Channel Estimation, Equalization, and Decoding (Adaptive Turbo Equalization)." Doctoral thesis, Norwegian University of Science and Technology, Faculty of Information Technology, Mathematics and Electrical Engineering, 2002. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-86.

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We address the problem of improving the throughput and the availability of digital communications in the High Frequency (HF, 3-30 MHz) band. In standardized military waveforms, the data is protected by an error-correcting code (ECC), and the code bits are shuffled by an interleaver and mapped onto a signal constellation for modulation onto a single carrier. Training sequences are multiplexed into the stream of transmitted symbols to aid the receiver in tracking the channel variations. The channel imposes severe time-varying intersymbol interference (ISI) as well as additive noise. Conventional receivers for such a system would first perform adaptive equalization (to mitigate the ISI) and symbol demapping, deinterleave the received code bits, and finally perform decoding, where the redundancy of the ECC is used to make high-quality decisions on the transmitted data bits even when bit errors have been introduced by the channel. Such a receiver is suboptimal because the equalizer does not make use of the redundancy introduced by the ECC, and is outperformed by an iterative scheme called turbo equalization. In turbo equalization, a.k.a. iterative equalization and decoding, soft information on the code bits is fed back from the decoder to the equalizer in an iterative fashion, and by performing the equalization and decoding tasks several times the bit error rates become significantly smaller than for a conventional “single-pass” receiver. Since we are dealing with an unknown time-varying channel, we must also perform channel estimation. We include channel estimation in the iterative loop of the turbo equalizer, using soft information fed back from the decoder as “training sequences” between the ordinary transmitted training sequences. Then, the receiver performs iterative channel estimation, equalization, and decoding, which can also be called adaptive turbo equalization. We have proposed a receiver using adaptive turbo equalization, and performed simulations using the MIL-STD-188-110 waveform at 2400 bps, transmitted over an ITU-R poor channel (a commonly used channel to test HF modems). We find that the proposed receiver outperforms a conventional receiver by 2-3 dB in terms of required signal-to-noise ratio to achieve a certain bit error rate. In this dissertation, we give an introduction to the fields of HF communications and standardized HF waveforms, channel modelling, and turbo equalization. We present an analysis of measured channel data to motivate our research in turbo equalization. We then present our research contributions to the field of turbo equalization: A low-complexity soft-in soft-out equalizer for time-varying channels, a comparative study of channel estimation algorithms using soft information as the input signal, and an investigation of adaptive turbo equalization using a technique known as EXIT charts. Finally, we present our main practical result, which is our proposed receiver and the corresponding simulation results.

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15

Kutlu, Mehmet. "Kalman filtering approach to blind equalization." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1993. http://handle.dtic.mil/100.2/ADA276320.

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16

Gilboy, John Joseph. "Equalization of school funding in Montana." Thesis, Montana State University, 1996. http://etd.lib.montana.edu/etd/1996/gilboy/GilboyJ1996.pdf.

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In 1988 the First Judicial Court of Lewis and Clark County ruled that because of disparities in spending among districts and a heavy reliance on local property taxes, the school funding system in Montana did not provide an equal opportunity for education. The Montana State Legislature responded by passing House Bills 28 and 667 in attempts to reduce the reliance on local property taxes and to bring the expenditures among districts closer together. This thesis examines school budgets for a representative sample of 301 elementary districts and 118 high school districts for fiscal years 1989, 1991 and 1995. Districts are ranked by their general fund budget per pupil in each year. High spending districts (95th percentile) are then compared to low spending districts (5th percentile). The results indicate that spending disparities have diminished among both elementary and high school districts, and among most size groups as well. High spending districts, however, still commonly spend twice as much as low spending districts, far exceeding the 1.25 ratio which is the target of both federal regulations and the state's own program. Changes in state policy over this period first reduced and then increased district dependence on local property tax levies. When HB 28 was first implemented, the state picked up a larger share of budgets in most districts. Although much of the state's contribution was itself financed by property taxes, districts did not need to rely so much on their local levies. Between 1991 and 1995, however, state funding failed to keep pace with inflation and enrollment growth. The state also changed the rules governing district finances so that voter approval is often necessary. The result of these policies has been a growing reliance since 1991 on local mill levies, 'and increasing numbers of public votes on budget issues. These trends may run counter to the goal of equalization, while restraining overall spending.

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McGinty, Nigel, and nigel mcginty@defence gov au. "Reduced Complexity Equalization for Data Communication." The Australian National University. Research School of Information Sciences and Engineering, 1998. http://thesis.anu.edu.au./public/adt-ANU20050602.122741.

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Optimal decision directed equalization techniques for time dispersive communication channels are often too complex to implement. This thesis considers reduced complexity decision directed equalization that lowers complexity demands yet retains close to optimal performance. The first part of this dissertation consists of three reduced complexity algorithms based on the Viterbi Algorithm (VA) which are: the Parallel Trellis VA (PTVA); Time Reverse Reduced State Sequence Estimation (TR-RSSE); and Forward-Backward State Sequence Detection (FBSSD). The second part of the thesis considers structural modifications of the Decision Feedback Equalizer (DFE), which is a special derivative of the VA, specifically, optimal vector quantization for fractionally spaced DFEs, and extended stability regions for baud spaced DFEs using passivity analysis are investigated.¶ For a special class of sparse channels the VA can be decomposed over a number of independent parallel trellises. This decomposition will be called the Parallel Trellis Viterbi Algorithm and can have lower complexity than the VA yet it retains optimal performance. By relaxing strict sparseness constraints on the channel a sub-optimal approach is proposed which keeps complexity low and obtains good performance.¶ Reduced State Sequence Estimation (RSSE) is a popular technique to reduce complexity. However, its deficiency can be the inability to adequately equalize non-minimum phase channels. For channels that have energy peaks in the tail of the impulse response (post-cursor dominant) RSSE's complexity must be close to the VA or performance will be poor. Using a property of the VA which makes it invariant to channel reversal, TR-RSSE is proposed to extend application of RSSE to post-cursor dominant channels.¶ To further extend the class of channels suitable for RSSE type processing, FBSSD is suggested. This uses a two pass processing method, and is suited to channels that have low energy pre and post-cursor. The first pass generates preliminary estimates used in the second pass to aid the decision process. FBSSD can range from RSSE to TR-RSSE depending on parameter settings.¶ The DFE is obtained when the complexity of RSSE is minimized. Two characterizing properties of the DFE, which are addressed in this thesis, are feedback and quantization. A novel fractionally spaced (FS) DFE structure is presented which allows the quantizer to be generalized relative to the quantizer used in conventional FS-DFEs. The quantizer can be designed according to a maximum a posteriori criterion which takes into account a priori statistical knowledge of error occurrences. A radically different quantizer can be obtained using this technique which can result in significant performance improvements.¶ Due to the feedback nature of the DFE a form of stability can be considered. After a decision error occurs, a stable DFE will, after some finite time and in the absence of noise, operate error free. Passivity analysis provides sufficient conditions to determine a class of channels which insures a DFE will be stable. Under conditions of short channels and small modulation alphabets, it is proposed that conventional passivity analysis can be extended to account for varying operator gains, leading to weaker sufficient conditions for stability (larger class of channels).

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Tam, Hing Sang Derek. "Adaptive equalization for serial digital interface." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0010/MQ34140.pdf.

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Xu, Yang. "Equalization algorithms for ADSL DMT system." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0004/MQ36756.pdf.

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Schrader, Johan Hendrik Rutger. "Wireline equalization using pulse-width modulation." Enschede : University of Twente [Host], 2007. http://doc.utwente.nl/58036.

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Çiftçi, Mahmut. "Channel equalization for chaotic communications systems." Diss., Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/15464.

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Leon, Wing Seng. "Equalization and estimation for fading channels." Thesis, University of Canterbury. Electrical and Electronic Engineering, 2003. http://hdl.handle.net/10092/6039.

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The main contribution of this thesis is the development of high performing, reduced complexity receivers for fading channels. Three different receiver structures are proposed and they operate without the need of any channel statistics. First, a double filtering receiver for systems employing DPSK modulation on channels with small delay and Doppler spreads is presented. BER performance results obtained via simulations and analysis show that the proposed structure outperforms the conventional matched filtering DPSK detector. Second, a polynomial predictor based sequence detector for flat-fading channels is presented. The receiver consists of a bank of polynomial least squares FIR predictors. The proposed receiver is not restricted only to systems using constant envelope modulation schemes. Analytical and simulated BER results are presented. In some cases, the proposed receiver performs only a few dB worse than an MLSE receiver with known channel statistics. Third, a sequence detector employing the polynomial based GRLS channel estimator is presented. The GRLS estimator is a generalization of the standard RLS algorithm and it requires a state space model of the channel to operate. It is shown that by using a polynomial or t-power series of the channel coefficients, a justifiable state space model may be derived without the need for any channel statistics. An analytical technique to evaluate the tracking performance of the GRLS estimator is also presented. The new analytical method may also be applied to the standard RLS algorithm with improved results. Simulated and analytical results show that in some cases, the tracking performances of the proposed channel estimator is almost as good as that of an optimal Kalman based channel estimator. BER results also indicate that the sequence detector using the proposed GRLS estimator performs just as well as one using a Kalman based estimator.

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COSTA, BERNARDO RODRIGUES DA. "CHANNEL EQUALIZATION IN BLOCK TRANSMISSION SYSTEMS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2007. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=11845@1.

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PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO
A necessidade pela transmissão em altas taxas, por exemplo nos novos sistemas de TV Digital de alta definição, telefonia celular de terceira e quarta geração, DSL e etc, trazem consigo um problema: o aumento dos múltiplos percursos no canal de comunicações (principalmente nas interfaces áereas), dando origem ao fenômeno de interferência intersimbólica (IES). Este trabalho investiga o desempenho de sistemas de transmissão com uma única portadora (SC ou single-carrier) com equalização linear no domínio da freqüência. Diferentes algoritmos recursivos são apresentados para implementar estes filtros FIR. Além dos equalizadores lineares, uma estrutura não-linear é introduzida, onde decisões passadas do decisor de mínima distância são utilizadas para mitigar os efeitos da IES na detecção dos símbolos subseqüentes. Este arranjo é conhecido como equalização/filtragem com decisões realimentadas (DFE ou Decision Feedback Equalizers). Por último, os resultados obtidos com o sistema SC nas diferentes configurações de filtragem na recepção são comparados com os resultados do já estabelecido sistema OFDM. A transmissão OFDM se dá com múltiplas portadoras, onde as freqüências das sub-portadoras são ortogonais entre si, permitindo que a informação seja enviada de forma paralela. Resultados mostram que os sistemas SC-FDE tem desempenho superior aos sistemas OFDM.
The demand for high rate transmission systems, for example in HDTV, third and fourth generation cellular telephony, DSL and so on, causes the rise of a problem: The multipath communications channel (specially in wireless communications), which leads to intersymbol interference phenomenon (ISI). The present work investigates the performance of single-carrier (SC) transmission systems with frequency-domain linear equalization. Different recursive algorithms are presented in order to implement these FIR filters. Besides the linear equalizers, a non- linear structure is introduced, where the past decisions made by the detectors are used to mitigate the effect of ISI on the detection of the forthcoming symbols. This set is known as Decision Feedback Equalizers (DFE). Finally, the results of the aforementioned systems are compared to the well-known OFDM. OFDM transmission relies on sub-carriers, frequency orthogonal to each other, in which the data is sent in a parallel basis. The results obtained show that SC- FDE systems outperform OFDM systems.

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Gurrapu, Omprakash. "Adaptive filter algorithms for channel equalization." Thesis, Högskolan i Borås, Institutionen Ingenjörshögskolan, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:hb:diva-19219.

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Equalization techniques compensate for the time dispersion introduced bycommunication channels and combat the resulting inter-symbol interference (ISI) effect.Given a channel of unknown impulse response, the purpose of an adaptive equalizer is tooperate on the channel output such that the cascade connection of the channel and theequalizer provides an approximation to an ideal transmission medium. Typically,adaptive equalizers used in digital communications require an initial training period,during which a known data sequence is transmitted. A replica of this sequence is madeavailable at the receiver in proper synchronism with the transmitter, thereby making itpossible for adjustments to be made to the equalizer coefficients in accordance with theadaptive filtering algorithm employed in the equalizer design. This type of equalization isknown as Non-Blind equalization. However, in practical situations, it would be highlydesirable to achieve complete adaptation without access to a desired response. Clearly,some form of Blind equalization has to be built into the receiver design. Blind equalizerssimultaneously estimate the transmitted signal and the channel parameters, which mayeven be time-varying. The aim of the project is to study the performance of variousadaptive filter algorithms for blind channel equalization through computer simulations.
Uppsatsnivå: D

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Wesel, Richard Dale. "Adaptive equalization for modem constellation identification." Thesis, Massachusetts Institute of Technology, 1989. http://hdl.handle.net/1721.1/29857.

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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1989.
Includes bibliographical references (leaves 77-78).
by Richard Dale Wesel.
M.S.

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Rice, Michael, Md Shah Afran, and Mohammad Saquib. "MMSE Equalization for Aeronautical Telemetry Channels." International Foundation for Telemetering, 2014. http://hdl.handle.net/10150/577447.

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ITC/USA 2014 Conference Proceedings / The Fiftieth Annual International Telemetering Conference and Technical Exhibition / October 20-23, 2014 / Town and Country Resort & Convention Center, San Diego, CA
This paper presents performance analysis of the minimum mean squared error (MMSE) equalizers applied to aeronautical telemetry channels. The challenge for equalizing received samples of the modulated signal lies in the fact that the underlying continuous-time SOQPSK-TG waveform is not wide-sense stationary. However it is assumed so in order to meet real-time implementation requirements. Two approximations of the autocorrelation function of the SOQPSK-TG waveform are used for designing MMSE equalizers. Their performance are investigated against the zero forcing equalizer for measured aeronautical telemetry channels.

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Kurak, Charles W. Jr. "Adaptive Histogram Equalization, a Parallel Implementation." UNF Digital Commons, 1990. http://digitalcommons.unf.edu/etd/260.

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Adaptive Histogram Equalization (AHE) has been recognized as a valid method of contrast enhancement. The main advantage of AHE is that it can provide better contrast in local areas than that achievable utilizing traditional histogram equalization methods. Whereas traditional methods consider the entire image, AHE utilizes a local contextual region. However, AHE is computationally expensive, and therefore time-consuming. In this work two areas of computer science, image processing and parallel processing, are combined to produce an efficient algorithm. In particular, the AHE algorithm is implemented with a Multiple-Instruction-Multiple-Data (MIMD) parallel architecture. It is proposed that, as MIMD machines become more powerful and prevalent, this methodology can be applied to not only this particular algorithm, but also to many others in its class.

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Shukla, Parveen Kumar. "Adaptive equalization of fading radio channels." Thesis, Imperial College London, 1989. http://hdl.handle.net/10044/1/47660.

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Liu, Yizhou. "ELECTRICAL EQUALIZATION FOR MULTIMODE FIBER SYSTEMS." Miami University / OhioLINK, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=miami1484004535118825.

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Zhang, Wancheng. "Robust equalization of multichannel acoustic systems." Thesis, Imperial College London, 2010. http://hdl.handle.net/10044/1/5882.

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In most real-world acoustical scenarios, speech signals captured by distant microphones from a source are reverberated due to multipath propagation, and the reverberation may impair speech intelligibility. Speech dereverberation can be achieved by equalizing the channels from the source to microphones. Equalization systems can be computed using estimates of multichannel acoustic impulse responses. However, the estimates obtained from system identification always include errors; the fact that an equalization system is able to equalize the estimated multichannel acoustic system does not mean that it is able to equalize the true system. The objective of this thesis is to propose and investigate robust equalization methods for multichannel acoustic systems in the presence of system identification errors. Equalization systems can be computed using the multiple-input/output inverse theorem or multichannel least-squares method. However, equalization systems obtained from these methods are very sensitive to system identification errors. A study of the multichannel least-squares method with respect to two classes of characteristic channel zeros is conducted. Accordingly, a relaxed multichannel least- squares method is proposed. Channel shortening in connection with the multiple- input/output inverse theorem and the relaxed multichannel least-squares method is discussed. Two algorithms taking into account the system identification errors are developed. Firstly, an optimally-stopped weighted conjugate gradient algorithm is proposed. A conjugate gradient iterative method is employed to compute the equalization system. The iteration process is stopped optimally with respect to system identification errors. Secondly, a system-identification-error-robust equalization method exploring the use of error models is presented, which incorporates system identification error models in the weighted multichannel least-squares formulation.

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Korst-Fagundes, Bruno Carleton University Dissertation Engineering Electronics. "Acoustical equalization at multiple listening positions." Ottawa, 1995.

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Kurkoski, Brian M. "Algorithms and schedules for turbo equalization /." Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2004. http://wwwlib.umi.com/cr/ucsd/fullcit?p3137222.

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Yao, Ning. "Iterative algorithms for channel estimation and equalization /." View abstract or full-text, 2005. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202005%20YAO.

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Mheidat, Hakam. "Channel Estimation and Equalization for Cooperative Communication." Thesis, University of Waterloo, 2006. http://hdl.handle.net/10012/2852.

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The revolutionary concept of space-time coding introduced in the last decade has demonstrated that the deployment of multiple antennas at the transmitter allows for simultaneous increase in throughput and reliability because of the additional degrees of freedom offered by the spatial dimension of the wireless channel. However, the use of antenna arrays is not practical for deployment in some practical scenarios, e. g. , sensor networks, due to space and power limitations.

A new form of realizing transmit diversity has been recently introduced under the name of user cooperation or cooperative diversity. The basic idea behind cooperative diversity rests on the observation that in a wireless environment, the signal transmitted by the source node is overheard by other nodes, which can be defined as "partners" or "relays". The source and its partners can jointly process and transmit their information, creating a "virtual antenna array" and therefore emulating transmit diversity.

Most of the ongoing research efforts in cooperative diversity assume frequency flat channels with perfect channel knowledge. However, in practical scenarios, e. g. broadband wireless networks, these assumptions do not apply. Frequency-selective fading and imperfect channel knowledge should be considered as a more realistic channel model. The development of equalization and channel estimation algorithms play a crucial element in the design of digital receivers as their accuracy determine the overall performance.

This dissertation creates a framework for designing and analyzing various time and frequency domain equalization schemes, i. e. distributed time reversal (D-TR) STBC, distributed single carrier frequency domain (D-SC-FDE) STBC, and distributed orthogonal frequency division multiplexing (D-OFDM) STBC schemes, for broadband cooperative communication systems. Exploiting the orthogonally embedded in D-STBCs, we were able to maintain low-decoding complexity for all underlying schemes, thus, making them excellent candidates for practical scenarios, such as multi-media broadband communication systems.

Furthermore, we propose and analyze various non-coherent and channel estimation algorithms to improve the quality and reliability of wireless communication networks. Specifically, we derive a non-coherent decoding rule which can be implemented in practice by a Viterbi-type algorithm. We demonstrate through the derivation of a pairwise error probability expression that the proposed non-coherent detector guarantees full diversity. Although this decoding rule has been derived assuming quasi-static channels, its inherent channel tracking capability allows its deployment over time-varying channels with a promising performance as a sub-optimal solution. As a possible alternative to non-coherent detection, we also investigate the performance of mismatched-coherent receiver, i. e. , coherent detection with imperfect channel estimation. Our performance analysis demonstrates that the mismatched-coherent receiver is able to collect the full diversity as its non-coherent competitor over quasi-static channels.

Finally, we investigate and analyze the effect of multiple antennas deployment at the cooperating terminals assuming different relaying techniques. We derive pairwise error probability expressions quantifying analytically the impact of multiple antenna deployment at the source, relay and/or destination terminals on the diversity order for each of the relaying methods under consideration.

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Lim, Sze Chie (Felicia). "Robust multichannel equalization for blind speech dereverberation." Thesis, Imperial College London, 2016. http://hdl.handle.net/10044/1/39566.

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Acoustic reverberation arises from the reflection of sound waves within an enclosed space. It is generally desirable in music reproduction but can be detrimental to speech-related applications. For the human listener, while the early reflections help to improve speech intelligibility, the late reflections have been shown to impair perceived speech quality. For speech processing technologies such as automatic speech recognizers, reverberation reduces accuracy and performance. Dereverberation is therefore an important research topic with interest driven by increasing availability of communication devices and consumer demand. One approach to dereverberation computes a set of equalizing filters that are used to perform the dereverberation processing, given multichannel inputs and estimates of the acoustic impulse responses (AIRs) between the source signal and microphones. However, estimation errors are inevitable in practice and therefore robust channel equalizers are required. This thesis aims to develop such robust algorithms in a manner that is desirable specifically for speech dereverberation. The framework of channel shortening is used, having been previously shown to give promising results. Subband approaches are also investigated to reduce the computational complexity and achieve finer control of dereverberation in separate frequency bands. A second approach to dereverberation steers the look direction of beamformers towards the source. Reverberant sounds from other directions are treated as noise and accordingly suppressed. The motivation behind beamformer design and channel equalization is similar and in this work, a unified framework termed MINTFormer is proposed. The aim is to combine the robustness of beamformers with the potentially perfect dereverberation ability that can be achieved by channel equalization approaches.

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Jain, Vijay. "A new sufficient-order blind equalization scheme." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape8/PQDD_0006/MQ43652.pdf.

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Yakoubian, Jeffrey Scott. "Adaptive histogram equalization for mammographic image processing." Thesis, Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/16387.

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Monfet, Frederic. "Turbo equalization using frequency-domain shortening filter." Thesis, McGill University, 2006. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=99527.

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Turbo equalization provides excellent performance but the complexity associated with this technique prohibits its use in application with severe inter-symbol interference (ISI) channel; the turbo equalizer complexity increases exponentially with the length of the channel impulse response (CIR) and the modulation level used for data transmission. In this work, a combined frequency-domain pre-equalizer with a turbo equalizer is proposed in an attempt to reduce the length of the CIR used by the turbo equalizer and hence the complexity of the receiver. The optimum selection of coefficients of the frequency-domain pre-equalizer and desired channel impulse response is discussed. With the proposed receiver, the complexity of the turbo equalizer can be controlled by pre-selecting the length of the desired channel impulse response. This complexity reduction is achieved at the cost of an increase in the noise level, which degrades the performance. The effect on the performance of such pre-equalizer is studied analytically. The overall performance of the proposed receiver for different length of the desired channel impulse response is studied via analytical comparison and simulation. Simulation results on performance in various frequency-selective fading channels indicate a substantial performance gain when compared to a conventional feed-forward equalizer (FFE) plus decision-feedback equalizer (DFE) receiver. Finally, in cases where a large alphabet is used for modulation, the reduced-search BCJR (Bahl, Cocke, Jelinek, and Raviv) [1] algorithm is utilized in the proposed receiver to further reduce the complexity of the receiver.

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Lambotharan, Sangarapillai. "Algorithms and structures for adaptive blind equalization." Thesis, Imperial College London, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.268038.

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Adnan, Rubyet. "Blind Equalization for Tomlinson-Harashima Precoded Systems." Thesis, University of Canterbury. Electrical and Computer Engineering, 2007. http://hdl.handle.net/10092/1130.

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At a communications receiver the observed signal is a corrupted version of the transmitted signal. This distortion in the received signal is due to the physical characteristics of the channel, including multipath propagation, the non-idealities of copper wires and impulse noise. Equalization is a process to combat these distortions in order to recover the original transmitted signal. Roughly stated, the equalizer tries to implement the inverse transfer function of the channel while taking into account the channel noise. The equalizer parameters can be tuned to this inverse transfer function using an adaptive algorithm. In many cases, the algorithm uses a training sequence to drive the equalizer parameters to the optimum solution. But, for time-varying channels or multiuser channels the use of a training sequence is inefficient in terms of bandwidth, as bandwidth is wasted due to the periodic re-transmission of the training sequence. A blind equalization algorithm is a practical method to eliminate this training sequence. An equalizer adapted using a blind algorithm is a key component of a bandwidth efficient receiver for broadcast and point-to-multipoint communications. The initial convergence performance of a blind adaptive equalizer depends on the higher-order statistics of the transmitted signal. In modern digital systems, Tomlinson-Harashima precoding (THP) is often used for signal shaping and to mitigate the error propagation problem of a decision feedback equalizer (DFE). The concept of THP comes from pre-equalization. In fact, it is a nonlinear form of pre-equalization, which bounds the higher-order statistics of the transmitted signal. But, THP and blind equalization are often viewed as incompatible equalization techniques. In this research, we give multiple scenarios where blind equalization of a THP-encoded signal might arise. With this motivation we set out to answer the question, can a blind equalizer successfully acquire a THP-encoded signal? We investigate the combination of a Tomlinson-Harashima precoder on the transmitter side and a blind equalizer on the receiver side. By bounding the kurtosis of the THP-encoded signal, we show that THP actually aids the initial convergence of blind equalization. We find that, as the symbol constellation size increases, the THP-encoded signal kurtosis approaches that of a uniform distribution, not a Gaussian. We investigate the compatibility of blind equalization with THP-encoded signals for both SISO and MIMO systems. In a SISO system, conventional blind algorithms can be used to counter the distortions introduced in the received signal. However, in a MIMO system with multiple users, the other users act as interferers on the desired user's signal. Hence, modified blind algorithms need to be applied to mitigate these interferers. For both SISO and MIMO systems, we show that the THP encoder ensures that the signal distribution approaches a non-Gaussian distribution. Using Monte Carlo simulations, we study the effects of Tomlinson-Harashima precoding on the performance of Bussgang-type blind algorithms and verify our theoretical analysis. The major contributions of this thesis are: • A demonstration that a blind equalizer can successfully acquire a THP-encoded signal for both SISO and MIMO systems. We show that THP actually aids blind equalization, as it ensures that the transmitted signal is non-Gaussian. • An analytical quantification of the effects of THP on the transmitted signal statistics. We derive a novel bound on the kurtosis of the THP-encoded signal. • An extension of the results from a single-user SISO scenario to multiple users and a MIMO scenario. We demonstrate that our bound and simulated results hold for these more general cases. Through our work, we have opened the way for a novel application of training sequence-less equalization: to acquire and equalize THP-encoded signals. Using our proposed system, periodic training sequences for a broadcast or point-to-multipoint system can be avoided, improving the bandwidth efficiency of the transceiver. Future modem designs with THP encoding can make use of our advances for bandwidth efficient communication systems.

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Clark, Martin Vivian. "Diversity and equalization in digital cellular radio." Thesis, University of Canterbury. Department of Electrical and Electronic Engineering, 1992. http://hdl.handle.net/10092/3278.

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This thesis analyses and quantifies the performance of a range of important diversity combining receivers operating in a digital cellular radio environment with linear modulation, frequency-selective Rayleigh multipath fading, and co-channel interference (CO). The following three types of diversity receiver are treated: (i) the maximum likelihood receiver (the best-possible receiver); (U) optimum linear receivers; and (iii) memoryless linear combining receivers with and without post-combiner equalization. The frequency-selective multipath fading channels are statistically characterized by a delay spectrum with an associated root-mean-square delay spread. The co-channel interference on each diversity branch is assumed to be composed of (i) a small number (possibly zero) of dominant interferers; and (ii) a large number of weak interferer; the sum of which is modelled as independent gaussian noise, or noise-like CCI. The potential bit-error-rate (BER) performance of the maxim likelihood receiver is analysed in the case of noise-like CCI alone (i.e., no dominant interferers), using the matched filter bound. The thesis presents a general. exact, and totally analytical solution of the matched filter bound on BER performance taken over the ensemble of multipath channel responses. The key to the solution is the application of the Karhunen-Loeve representation of these channel responses. The BER performance of an optimum linear receiver. optimized according to the minimum mean-square error (MMSE) criterion, is estimated using (i) a range of BER computation methods, including Metzger's algorithm and Saltzberg's bound; and (ii) random generation of the channel responses, using the efficient Karhunen-Loeve method. Five sub-classes of the memoryless linear combining receiver, with and without post-combiner equalization, are studied. They include arrangements of the following combining and equalization schemes: maximal ratio combining, maximal power combining, MMSE combining, MMSE linear equalization, and ideal intersymbol interference (lSI) cancellation. The BER performances of the five receiver sub-classes are estimated using similar techniques to those used for the optimum (MMSE) linear receiver. For quaternary phase shift keying (QPSK) and all the above receivers, the thesis presents a set of numerical results that show the influence of the diversity order. the number of dominant interferers (set to zero for matched filter bounds), the delay spectrum shape, the delay spread, and the signal-to-interference ratio. This extensive set of data shows the power of diversity and equalization at combating lSI and eel in frequency-selective fading environments, and important tradeoffs between receiver performance and complexity.

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Saleem, Sajid. "Frequency-domain equalization for continuous phase modulation." Diss., Georgia Institute of Technology, 2013. http://hdl.handle.net/1853/50391.

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Continuous phase modulation~(CPM) is a non-linear, constant-envelope modulation scheme with memory, known for its bandwidth and power efficiency. Multi-h CPM uses multiple modulation indices in successive symbol intervals to improve the error performance as compared to single-h CPM~(basic CPM that utilizes only a single modulation index). One of the major applications of multi-h CPM is in aeronautical telemetry systems. Modern aeronautical devices host an increasing number of sensors, which can transmit flight testing data to the ground station. However, this excess data transfer increases the intersymbol interference, and thus channel equalization is required at the receiver. The objective of our research is to propose low-complexity frqeuency-domain equalization~(FDE) techniques for multi-h CPM waveforms. For a modulation scheme with memory, such as CPM, the cyclic constraint on the FDE block necessitates the use of an extra segment of symbols, called intrafix or tail segment. We have used very simple geometric arguments to derive upper and lower bounds on the length of the intrafix in terms of the parameters of the modulation scheme and the Frobenius number. It is concluded that the length of the intrafix for multi-h CPM schemes is typically shorter than those required for single-h modulation schemes. We propose two receiver architectures; one uses a matched filter front end, while the other utilizes a fractional sampling front end. Various simplifications are proposed for each architecture, and the trade-off between receiver complexity and performance is analyzed and verified through detailed simulation studies.

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Jalali, Sammuel. "Wireless Channel Equalization in Digital Communication Systems." Scholarship @ Claremont, 2012. http://scholarship.claremont.edu/cgu_etd/42.

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Our modern society has transformed to an information-demanding system, seeking voice, video, and data in quantities that could not be imagined even a decade ago. The mobility of communicators has added more challenges. One of the new challenges is to conceive highly reliable and fast communication system unaffected by the problems caused in the multipath fading wireless channels. Our quest is to remove one of the obstacles in the way of achieving ultimately fast and reliable wireless digital communication, namely Inter-Symbol Interference (ISI), the intensity of which makes the channel noise inconsequential. The theoretical background for wireless channels modeling and adaptive signal processing are covered in first two chapters of dissertation. The approach of this thesis is not based on one methodology but several algorithms and configurations that are proposed and examined to fight the ISI problem. There are two main categories of channel equalization techniques, supervised (training) and blind unsupervised (blind) modes. We have studied the application of a new and specially modified neural network requiring very short training period for the proper channel equalization in supervised mode. The promising performance in the graphs for this network is presented in chapter 4. For blind modes two distinctive methodologies are presented and studied. Chapter 3 covers the concept of multiple "cooperative" algorithms for the cases of two and three cooperative algorithms. The "select absolutely larger equalized signal" and "majority vote" methods have been used in 2-and 3-algoirithm systems respectively. Many of the demonstrated results are encouraging for further research. Chapter 5 involves the application of general concept of simulated annealing in blind mode equalization. A limited strategy of constant annealing noise is experimented for testing the simple algorithms used in multiple systems. Convergence to local stationary points of the cost function in parameter space is clearly demonstrated and that justifies the use of additional noise. The capability of the adding the random noise to release the algorithm from the local traps is established in several cases.

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Greenfield, Richard Glentworth. "Application of digital techniques to loadspeaker equalization." Thesis, University of Essex, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.292175.

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Song, Sanquan 1980. "Fractionally spaced equalization for high-speed links." Thesis, Massachusetts Institute of Technology, 2011. http://hdl.handle.net/1721.1/64588.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.
Cataloged from PDF version of thesis.
Includes bibliographical references (p. 115-119).
As high-speed links enter the multi-Gb/s era, equalization and clock recovery designs become much more challenging. For conventional links, these two loops are separate with different performance metrics, resulting in sub-optimal performance. Fractionally spaced equalization (FSE) inherently unifies these two functions, and therefore is proposed for joint equalization and synchronization in this thesis. At the system level, this thesis introduces new adaptation techniques for both mesochronous and plesiochronous applications. For mesochronous systems, the divergence issue of the low-cost sign-sign least-mean-square (SSLMS) adaptive algorithm is solved by using update conditioning to effectively increase the quantization resolution. For plesiochronous systems, a digitally-controlled bit-skipping scheme is proposed for frequency offset compensation. At the circuit level, the voltage-time conversion technique is redesigned to build highspeed, linear and energy-efficient FSE filter taps, which are scalable to advanced technology nodes. All the information is processed by linear current integration, with all integrated currents independent of the channel voltages, avoiding the non-linear voltage-current transformation. Based on different voltage-to-time converter designs, two proof-of-concept FSE implementations have been fabricated in a 90-nm CMOS process. The first implementation is a 2-way interleaved 2-tap FSE, operating at 4.0 Gb/s, with 2.0 pJ/bit energy-efficiency and 4.3 bits of linearity, showing immunity to the sampling phase. Operating at higher rates (6.25 Gb/s), the second implementation is designed as a 4-way interleaved 2-tap FSE with a 1-tap DFE, which achieves 3.6 pJ/bit energy-efficiency and over 4.0 bits of linearity, demonstrating the convergence of the modified sign-sign least-mean-square (M-SSLMS) algorithm. A third implementation has been designed with on-chip coefficient adaptation loop and bit-skipping scheme for plesiochronous systems.
by Sanquan Song.
Ph.D.

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Barron, Richard J. (Richard John). "Channel equalization for self-synchronizing chaotic systems." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/38828.

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Lei, Qiang, and Michael Rice. "Iterative Equalization for SOQPSK in Multipath Fading." International Foundation for Telemetering, 2008. http://hdl.handle.net/10150/606195.

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ITC/USA 2008 Conference Proceedings / The Forty-Fourth Annual International Telemetering Conference and Technical Exhibition / October 27-30, 2008 / Town and Country Resort & Convention Center, San Diego, California
This paper investigates the application of iterative equalization techniques to overcome multipath fading for shaped offset QPSK (SOQPSK) in aeronautical telemetry. Two iterative equalization techniques for turbo encoded SOQPSK are presented. The first is the optimal-MAP turbo equalizer for OQPSK. The second equalizer is the adaptive decision feedback equalizer. Simulation shows that in the presence of frequency selective multipath typically encountered in aeronautical telemetry, both of these equalizers exhibit impressive performance.

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ABASCAL, CARLOS G. "ADAPTIVE EQUALIZATION OF A RADIO FREQUENCY AMPLIFIER." University of Cincinnati / OhioLINK, 2001. http://rave.ohiolink.edu/etdc/view?acc_num=ucin983391684.

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Mathew, Jerry George. "MIMO equalization." Thesis, 2005. http://hdl.handle.net/10413/2815.

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In recent years, space-time block co'des (STBC) for multi-antenna wireless systems have emerged as attractive encoding schemes for wireless communications. These codes provide full diversity gain and achieve good performance with simple receiver structures without the additional increase in bandwidth or power requirements. When implemented over broadband channels, STBCs can be combined with orthogonal frequency division multiplexing (OFDM) or single carrier frequency domain (SC-FD) transmission schemes to achieve multi-path diversity and to decouple the broadband frequency selective channel into independent flat fading channels. This dissertation focuses on the SC-FD transmission schemes that exploit the STBC structure to provide computationally cost efficient receivers in terms of equalization and channel estimation. The main contributions in this dissertation are as follows: • The original SC-FD STBC receiver that bench marks STBC in a frequency selective channel is limited to coherent detection where the knowledge of the channel state information (CSI) is assumed at the receiver. We extend this receiver to a multiple access system. Through analysis and simulations we prove that the extended system does not incur any performance penalty. This key result implies that the SC-FD STBC scheme is suitable for multiple-user systems where higher data rates are possible. • The problem of channel estimation is considered in a time and frequency selective environment. The existing receiver is based on a recursive least squares (RLS) adaptive algorithm and provides joint equalization and interference suppression. We utilize a system with perfect channel state information (CSI) to show from simulations how various design parameters for the RLS algorithm can be selected in order to get near perfect CSI performance. • The RLS receiver has two modes of operation viz. training mode and direct decision mode. In training mode, a block of known symbols is used to make the initial estimate. To ensure convergence of the algorithm a re-training interval must be predefined. This results in an increase in the system overhead. A linear predictor that utilizes the knowled~e of the autocorrelation function for a Rayleigh fading channel is developed. The predictor is combined with. the adaptive receiver to provide a bandwidth efficient receiver by decreasing the training block size.· The simulation results show that the performance penalty for the new system is negligible. • Finally, a new Q-R based receiver is developed to provide a more robust solution to the RLS adaptive receiver. The simulation results clearly show that the new receiver outperforms the RLS based receiver at higher Doppler frequencies, where rapid channel variations result in numerical instability of the RLS algorithm. The linear predictor is also added to the new receiver which results in a more robust and bandwidth efficient receiver.
Thesis (M.Sc.Eng.)-University of KwaZulu-Natal, Durban, 2005.

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Chow, Y. F., and 周永富. "Acoustical Spatial Equalization." Thesis, 1994. http://ndltd.ncl.edu.tw/handle/77119023421739345465.

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碩士
國立交通大學
電信研究所
82
This thesis describes a general theoretical basis for the design of multichannel acoustical spatial equalizers to provide a good acoustical sound field. We aim to equalize both the response of the loudspeakers and the listening room as well as to cancel the acoustic crosstalk. The work presented is applied first to the single-channel, two-channel and then extended to the multi-channel case. Several methods to obtain the solution of the equalization system are provided. Extension of acoustical equalization over a listening area is also discussed. Because the room impulse response varies with location, the issue of mismatched equalization based on matrix perturbation theory is addressed. Multi-channel equalization using adaptive filters offers an attractive solution to the reduction of computation and complexity, we will use the LMS approach to design the digital filters. Finally, computer simulations will justify the algorithms for acoustical spatial equalization.

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Dissertations / Theses on the topic 'Equalization'

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