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1
Mitchell, Daniel Allan. "Interference Mitigation in Radio Astronomy." Thesis, The University of Sydney, 2004. http://hdl.handle.net/2123/693.
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This thesis investigates techniques and algorithms for mitigating radio frequency interference (RFI) affecting radio astronomy observations. In the past radio astronomy has generally been performed in radio-quiet geographical locations and unused parts of the radio spectrum, including small protected frequency bands. The increasing use of the entire spectrum and global transmitters such as satellites are forcing the astronomy community to begin implementing active interference cancelling. The amount of harmful interference affecting observations will also increase as future instruments such as the Square Kilometre Array (SKA) are required to use larger bandwidths to reach up to 100 times the current sensitivity levels, and as spectral line observations require observing in bands licensed to other spectrum users. Particular attention is paid to interference cancellation algorithms which make use of reference beams. This has proven to be successful in removing interference from the contaminated astronomical data. Reference antenna cancellers are closely analysed, leading to filters and techniques that can offer improved RFI excision for some important applications. It is shown that pre- and post-correlation reference antenna cancellers give similar results, and an important aspect of the cancellers is the use of a second reference signal when the reference interference-to-noise ratio is low. These modified filters can theoretically offer infinite interference suppression in the voltage domain, equivalent to that of post-correlation interference cancellers, and their internal structure can offer an understanding of the residual RFI and added receiver noise components of a variety of reference antenna techniques. The effect of variable geometric delays is also considered and various filters are compared as a function of the geometric fringe rate.
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Mitchell, Daniel Allan. "Interference Mitigation in Radio Astronomy." University of Sydney. Physics, 2004. http://hdl.handle.net/2123/693.
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This thesis investigates techniques and algorithms for mitigating radio frequency interference (RFI) affecting radio astronomy observations. In the past radio astronomy has generally been performed in radio-quiet geographical locations and unused parts of the radio spectrum, including small protected frequency bands. The increasing use of the entire spectrum and global transmitters such as satellites are forcing the astronomy community to begin implementing active interference cancelling. The amount of harmful interference affecting observations will also increase as future instruments such as the Square Kilometre Array (SKA) are required to use larger bandwidths to reach up to 100 times the current sensitivity levels, and as spectral line observations require observing in bands licensed to other spectrum users. Particular attention is paid to interference cancellation algorithms which make use of reference beams. This has proven to be successful in removing interference from the contaminated astronomical data. Reference antenna cancellers are closely analysed, leading to filters and techniques that can offer improved RFI excision for some important applications. It is shown that pre- and post-correlation reference antenna cancellers give similar results, and an important aspect of the cancellers is the use of a second reference signal when the reference interference-to-noise ratio is low. These modified filters can theoretically offer infinite interference suppression in the voltage domain, equivalent to that of post-correlation interference cancellers, and their internal structure can offer an understanding of the residual RFI and added receiver noise components of a variety of reference antenna techniques. The effect of variable geometric delays is also considered and various filters are compared as a function of the geometric fringe rate.
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Picciolo, Michael L. "Robust Adaptive Signal Processors." Diss., Virginia Tech, 2003. http://hdl.handle.net/10919/26993.
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Standard open loop linear adaptive signal processing algorithms derived from the least squares minimization criterion require estimates of the N-dimensional input interference and noise statistics. Often, estimated statistics are biased by contaminant data (such as outliers and non-stationary data) that do not fit the dominant distribution, which is often modeled as Gaussian. In particular, convergence of sample covariance matrices used in block processed adaptive algorithms, such as the Sample Matrix Inversion (SMI) algorithm, are known to be affected significantly by outliers, causing undue bias in subsequent adaptive weight vectors. The convergence measure of effectiveness (MOE) of the benchmark SMI algorithm is known to be relatively fast (order K = 2N training samples) and independent of the (effective) rank of the external interference covariance matrix, making it a useful method in practice for non-contaminated data environments. Novel robust adaptive algorithms are introduced here that perform superior to SMI algorithms in contaminated data environments while some retain its valuable convergence independence feature. Convergence performance is shown to be commensurate with SMI in non-contaminated environments as well. The robust algorithms are based on the Gram Schmidt Cascaded Canceller (GSCC) structure where novel building block algorithms are derived for it and analyzed using the theory of Robust Statistics. Coined M â cancellers after M â estimates of Huber, these novel cascaded cancellers combine robustness and statistical estimation efficiency in order to provide good adaptive performance in both contaminated and non-contaminated data environments. Additionally, a hybrid processor is derived by combining the Multistage Wiener Filter (MWF) and Median Cascaded Canceller (MCC) algorithms. Both simulated data and measured Space-Time Adaptive Processing (STAP) airborne radar data are used to show performance enhancements. The STAP application area is described in detail in order to further motivate research into robust adaptive processing.Ph. D.
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Zerguine, Azzedine. "Algorithms and structures for long adaptive echo cancellers." Thesis, Loughborough University, 1996. https://dspace.lboro.ac.uk/2134/22076.
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The main theme of this thesis is adaptive echo cancellation. Two novel independent approaches are proposed for the design of long echo cancellers with improved performance. In the first approach, we present a novel structure for bulk delay estimation in long echo cancellers which considerably reduces the amount of excess error. The miscalculation of the delay between the near-end and the far-end sections is one of the main causes of this excess error. Two analyses, based on the Least Mean Squares (LMS) algorithm, are presented where certain shapes for the transitions between the end of the near-end section and the beginning of the far-end one are considered. Transient and steady-state behaviours and convergence conditions for the proposed algorithm are studied. Comparisons between the algorithms developed for each transition are presented, and the simulation results agree well with the theoretical derivations. In the second approach, a generalised performance index is proposed for the design of the echo canceller. The proposed algorithm consists of simultaneously applying the LMS algorithm to the near-end section and the Least Mean Fourth (LMF) algorithm to the far-end section of the echo canceller. This combination results in a substantial improvement of the performance of the proposed scheme over both the LMS and other algorithms proposed for comparison. In this approach, the proposed algorithm will be henceforth called the Least Mean Mixed-Norm (LMMN) algorithm. The advantages of the LMMN algorithm over previously reported ones are two folds: it leads to a faster convergence and results in a smaller misadjustment error. Finally, the convergence properties of the LMMN algorithm are derived and the simulation results confirm the superior performance of this proposed algorithm over other well known algorithms.
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Sankaran, Sundar G. "Implementation and evaluation of echo cancellation algorithms." Thesis, This resource online, 1996. http://scholar.lib.vt.edu/theses/available/etd-02132009-172004/.
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Campbell, David Kemp. "Adaptive Beamforming Using a Microphone Array for Hands-Free Telephony." Thesis, Virginia Tech, 1999. http://hdl.handle.net/10919/31294.
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This thesis describes the design and implementation of a 4-channel microphone array that is an adaptive beamformer used for hands-free telephony in a noisy environment. The microphone signals are amplified, then sent to an A/D converter. The microprocessor board takes the data from the 4 channels and utilizes digital signal processing to determine the direction-of-arrival of the sources and create an output which 'steers' the microphone array to the desired look direction while trying to minimize the energy of interference sources and noise. All of the processing for this thesis will be done on a computer using MATLAB.
The MUSIC algorithm is used for direction finding in this thesis. It is shown to be effective in estimating direction-of-arrival for 1 speech source and 2 speech sources that are spaced fairly apart, with significant results down to a -5 dB SNR even. The MUSIC algorithm requires knowledge of the number of sources a priori, requiring an estimator for the number of sources. Though proposed estimators for the number of sources were examined, an effective estimator was not encountered for the case where there are multiple speech sources.
Beamforming methods are examined which utilize knowledge of the source direction-of-arrival from the MUSIC algorithm. The input is split into 6 subbands such that phase-steered beamforming would be possible. Two methods of phase-steered beamforming are compared in both narrowband and wideband scenarios, and it is shown that phase-steering the array to the desired source direction-of-arrival has about 0.3 dB better beamforming performance than the simple time-delay steered beamformer using no subbands.
As the beamforming solution is inadequate to achieve desired results, a generalized sidelobe canceler (GSC) is developed which incorporates a beamformer. The sidelobe canceler is evaluated using both NLMS and RLS adaptation. The RLS algorithm inherently gives better results than the NLMS algorithm, though the computational complexity renders the solution impractical for implementation with today's technology.
A testing setup is presented which involves a linear 4-microphone array connected to a DSP chip that collects the data. Tests were done using 1 speech source and a model of the car noise environment. The sidelobe canceler's performance using 6 subbands (phase-delay GSC) and using 1 band (time-delay GSC) with NLMS updating are compared. The overall SNR improvement is determined from the signal and noise input and output powers, with signal-only as the input and noise-only as the input to the GSC. The phase-delay GSC gives on average 7.4 dB SNR improvement while the time-delay GSC gives on average 6.2 dB SNR improvement.Master of Science
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Mackey, Richard Paul. "An asynchronous, single-chip, LMS based, adaptive fir echo canceller." Thesis, The University of Arizona, 1995. http://hdl.handle.net/10150/291387.
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An asynchronous, single-chip, high-speed communication adaptive echo canceller was developed during this research. Adaptation is based on the LMS algorithm with power-of-two convergence factor. Cancellation is performed by a 128-coefficient adaptive finite impulse response filter whose coefficients are updated every cycle. The LMS power-of-two update equations were modified to allow a pipelined implementation. Pipelining the adaptation and echo estimation operations enabled hardware minimization, a high sampling rate, and no increase in convergence time. The resulting circuit updates the filter coefficients and generates the output at a sampling rate greater than 205 kHz. The chip was designed using 0.8 mum CMOS standard cells. The single-chip layout requires a die size of 9.25 mm by 7.25 mm.
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Yamazaki, Ken Carleton University Dissertation Engineering Electrical. "Convergence behaviour of a jointly-adaptive transversal and memory- based echo canceller." Ottawa, 1989.
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Brophy, Sean G. (Sean Gregory) Carleton University Dissertation Engineering Electrical. "Synchronization and its effect on adaptive echo canceller performance in the digital subscriber loop environment." Ottawa, 1985.
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Karim, Md Anisul. "Weighted layered space-time code with iterative detection and decoding." Thesis, The University of Sydney, 2006. http://hdl.handle.net/2123/1095.
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Multiple antenna systems are an appealing candidate for emerging fourth-generation wireless networks due to its potential to exploit space diversity for increasing conveyed throughput without wasting bandwidth and power resources. Particularly, layered space-time architecture (LST) proposed by Foschini, is a technique to achieve a significant fraction of the theoretical capacity with a reasonable implementation complexity. There has been a great deal of challenges in the detection of space-time signal; especially to design a low-complexity detector, which can efficiently remove multi-layer interference and approach the interference free bound. The application of iterative principle to joint detection and decoding has been a promising approach. It has been shown that, the iterative receiver with parallel interference canceller (PIC) has a low linear complexity and near interference free performance. Furthermore, it is widely accepted that the performance of digital communication systems can be considerably improved once the channel state information (CSI) is used to optimize the transmit signal. In this thesis, the problem of the design of a power allocation strategy in LST architecture to simultaneously optimize coding, diversity and weighting gains is addressed. A more practical scenario is also considered by assuming imperfect CSI at the receiver. The effect of channel estimation errors in LST architecture with an iterative PIC receiver is investigated. It is shown that imperfect channel estimation at an LST receiver results in erroneous decision statistics at the very first iteration and this error propagates to the subsequent iterations, which ultimately leads to severe degradation of the overall performance. We design a transmit power allocation policy to take into account the imperfection in the channel estimation process. The transmit power of various layers is optimized through minimization of the average bit error rate (BER) of the LST architecture with a low complexity iterative PIC detector. At the receiver, the PIC detector performs both interference regeneration and cancellation simultaneously for all layers. A convolutional code is used as the constituent code. The iterative decoding principle is applied to pass the a posteriori probability estimates between the detector and decoders. The decoder is based on the maximum a posteriori (MAP) algorithms. A closed-form optimal solution for power allocation in terms of the minimum BER is obtained. In order to validate the effectiveness of the proposed schemes, substantial simulation results are provided.
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Karim, Md Anisul. "Weighted layered space-time code with iterative detection and decoding." School of Electrical & Information Engineering, 2006. http://hdl.handle.net/2123/1095.
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Master of Engineering (Research)Multiple antenna systems are an appealing candidate for emerging fourth-generation wireless networks due to its potential to exploit space diversity for increasing conveyed throughput without wasting bandwidth and power resources. Particularly, layered space-time architecture (LST) proposed by Foschini, is a technique to achieve a significant fraction of the theoretical capacity with a reasonable implementation complexity. There has been a great deal of challenges in the detection of space-time signal; especially to design a low-complexity detector, which can efficiently remove multi-layer interference and approach the interference free bound. The application of iterative principle to joint detection and decoding has been a promising approach. It has been shown that, the iterative receiver with parallel interference canceller (PIC) has a low linear complexity and near interference free performance. Furthermore, it is widely accepted that the performance of digital communication systems can be considerably improved once the channel state information (CSI) is used to optimize the transmit signal. In this thesis, the problem of the design of a power allocation strategy in LST architecture to simultaneously optimize coding, diversity and weighting gains is addressed. A more practical scenario is also considered by assuming imperfect CSI at the receiver. The effect of channel estimation errors in LST architecture with an iterative PIC receiver is investigated. It is shown that imperfect channel estimation at an LST receiver results in erroneous decision statistics at the very first iteration and this error propagates to the subsequent iterations, which ultimately leads to severe degradation of the overall performance. We design a transmit power allocation policy to take into account the imperfection in the channel estimation process. The transmit power of various layers is optimized through minimization of the average bit error rate (BER) of the LST architecture with a low complexity iterative PIC detector. At the receiver, the PIC detector performs both interference regeneration and cancellation simultaneously for all layers. A convolutional code is used as the constituent code. The iterative decoding principle is applied to pass the a posteriori probability estimates between the detector and decoders. The decoder is based on the maximum a posteriori (MAP) algorithms. A closed-form optimal solution for power allocation in terms of the minimum BER is obtained. In order to validate the effectiveness of the proposed schemes, substantial simulation results are provided.
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Ahmed, Mamun. "Adaptive Sub band GSC Beam forming using Linear Microphone-Array for Noise Reduction/Speech Enhancement." Thesis, Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, 2012. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-6174.
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This project presents the description, design and the implementation of a 4-channel microphone array that is an adaptive sub-band generalized side lobe canceller (GSC) beam former uses for video conferencing, hands-free telephony etc, in a noisy environment for speech enhancement as well as noise suppression. The side lobe canceller evaluated with both Least Mean Square (LMS) and Normalized Least Mean Square (NLMS) adaptation. A testing structure is presented; which involves a linear 4-microphone array connected to collect the data. Tests were done using one target signal source and one noise source. In each microphone’s, data were collected via fractional time delay filtering then it is divided into sub-bands and applied GSC to each of the subsequent sub-bands. The overall Signal to Noise Ratio (SNR) improvement is determined from the main signal and noise input and output powers, with signal-only and noise-only as the input to the GSC. The NLMS algorithm significantly improves the speech quality with noise suppression levels up to 13 dB while LMS algorithm is giving up to 10 dB. All of the processing for this thesis is implemented on a computer using MATLAB and validated by considering different SNR measure under various types of blocking matrix, different step sizes, different noise locations and variable SNR with noise.Mamun Ahmed E-mail: mamuncse99cuet@yahoo.com
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Gao, Ying. "A Digital Signal Processing Approach for Affective Sensing of a Computer User through Pupil Diameter Monitoring." FIU Digital Commons, 2009. http://digitalcommons.fiu.edu/etd/132.
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Recent research has indicated that the pupil diameter (PD) in humans varies with their affective states. However, this signal has not been fully investigated for affective sensing purposes in human-computer interaction systems. This may be due to the dominant separate effect of the pupillary light reflex (PLR), which shrinks the pupil when light intensity increases. In this dissertation, an adaptive interference canceller (AIC) system using the H∞ time-varying (HITV) adaptive algorithm was developed to minimize the impact of the PLR on the measured pupil diameter signal. The modified pupil diameter (MPD) signal, obtained from the AIC was expected to reflect primarily the pupillary affective responses (PAR) of the subject. Additional manipulations of the AIC output resulted in a processed MPD (PMPD) signal, from which a classification feature, PMPDmean, was extracted. This feature was used to train and test a support vector machine (SVM), for the identification of stress states in the subject from whom the pupil diameter signal was recorded, achieving an accuracy rate of 77.78%. The advantages of affective recognition through the PD signal were verified by comparatively investigating the classification of stress and relaxation states through features derived from the simultaneously recorded galvanic skin response (GSR) and blood volume pulse (BVP) signals, with and without the PD feature. The discriminating potential of each individual feature extracted from GSR, BVP and PD was studied by analysis of its receiver operating characteristic (ROC) curve. The ROC curve found for the PMPDmean feature encompassed the largest area (0.8546) of all the single-feature ROCs investigated. The encouraging results seen in affective sensing based on pupil diameter monitoring were obtained in spite of intermittent illumination increases purposely introduced during the experiments. Therefore, these results confirmed the benefits of using the AIC implementation with the HITV adaptive algorithm to isolate the PAR and the potential of using PD monitoring to sense the evolving affective states of a computer user.
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謝忠文. "Modular Adaptive Filter Design and Implementation of Ghost Canceler." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/42061718035590016898.
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尤閔生. "The Study of Adaptive Digital Array Noise Canceller." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/15875441277364341617.
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碩士逢甲大學
電機工程研究所
86
Active control of a single channel system produces only a zone of quit around the monitor microphone. For noise cancellation of a large space, the multichannel system is needed. Hence, in this research, we intend to use the array of microphones and loudspeakers to generate zones of quite. The performance of the single channel system has a great influence on the multichannel because the multichannel channel system is an extension of the single channel system. Usually time-domain filtered-x LMS algorithm in the single channel system does not have good perform ace, this is due to the real model is IIR filters. For the simplicity and the stability considerations, most researchers adapt the FIR filters.In order to have a good performance by using FIR filters, thousands of the filter coefficients may be used. Thsimakes the computation lad largely increased. In this research, we use the frequency-domain filtered-x LMS algorithm to reduce the computation effort. There are two methods in frequency -domain filtered-x LMS algorithm One is the constrained method. The other is the unconstrained method. The peroformace of the former algorithm is as same as the conventional time-domain filtered-x LMS algorithm. When the wrapround problem on the constrained method thas a little effect on the system, the performace of the unconstrained method is epual to of better than the constrained method and the time-domain algorithm.
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"Mixed-Mode Adaptive Ripple Canceller for Switching Regulators." Master's thesis, 2016. http://hdl.handle.net/2286/R.I.40787.
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abstract: State of art modern System-On-Chip architectures often require very low noise supplies without overhead on high efficiencies. Low noise supplies are especially important in noise sensitive analog blocks such as high precision Analog-to-Digital Converters, Phase Locked Loops etc., and analog signal processing blocks. Switching regulators, while providing high efficiency power conversion suffer from inherent ripple on their output. A typical solution for high efficiency low noise supply is to cascade switching regulators with Low Dropout linear regulators (LDO) which generate inherently quiet supplies. The switching frequencies of switching regulators keep scaling to higher values in order to reduce the sizes of the passive inductor and capacitors at the output of switching regulators. This poses a challenge for existing solutions of switching regulators followed by LDO since the Power Supply Rejection (PSR) of LDOs are band-limited. In order to achieve high PSR over a wideband, the penalty would be to increase the quiescent power consumed to increase the bandwidth of the LDO and increase in solution area of the LDO. Hence, an alternative to the existing approach is required which improves the ripple cancellation at the output of switching regulator while overcoming the deficiencies of the LDO.
This research focuses on developing an innovative technique to cancel the ripple at the output of switching regulator which is scalable across a wide range of switching frequencies. The proposed technique consists of a primary ripple canceller and an auxiliary ripple canceller, both of which facilitate in the generation of a quiet supply and help to attenuate the ripple at the output of buck converter by over 22dB. These techniques can be applied to any DC-DC converter and are scalable across frequency, load current, output voltage as compared to LDO without significant overhead on efficiency or area. The proposed technique also presents a fully integrated solution without the need of additional off-chip components which, considering the push for full-integration of Power Management Integrated Circuits, is a big advantage over using LDOs.Dissertation/Thesis
Masters Thesis Electrical Engineering 2016
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Cho, Jing Lun, and 卓經綸. "Robust Adaptive Array Beamforming Using Generalized Sidelobe Canceller." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/60007714589817918414.
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碩士國立臺灣大學
電信工程學研究所
90
Beamforming technique can suppress interference.But if there are some nonideal situations, beamforming technique can't suppress interference.For example,the antenna location has error, or some near field signals exist. We develop some algorithms to improve the performance of beamforming at steering vector error.In the other words,we develope robust beamforming to degrade the effect of sensor position error and near field signal. We primarily discuss beamforming with GSC architecture.This architecture is easy for implementation.In addition,we also develope blind beamforming with GSC architecure using cyclostationarity of signal.
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Wu, Chau-Ching, and 吳兆鈞. "Design and Implementation of An Asynchronous Adaptive Echo Canceller." Thesis, 1998. http://ndltd.ncl.edu.tw/handle/82328711319766075584.
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碩士國立成功大學
電機工程學系
86
Echo has disturbed us under many conditions for a long time. For example, it might occur in telephone or satellite communication system or in microphone-loudspeaker based system. The appearance of echo in these communication systems will reduce the communication quality and cause the user feel uncomfortable. Thus in this thesis, we developed a LMS algorithm and adaptive filter based asynchronous echo canceller to solve this problem. During the design procedure, we firstadopt a simplified LMS algorithm with power of two convergence factor to reduce the complexity of the algorithm. Following, we design the hardware structure using pipeline style. These two methods drastically reduce the requirement of hardware area. Additional, in order to improve the speed of the echo canceller, we adopt asynchronous design styleinstead of conventional synchronous style. Besides, the advantages of lower power consumptionand easy to extend and modify are also the reasonsthat appeal us. The whole design is simulated using Altera MAXPLUS II ver.8.0 software and it successfully fits into the EPF10K100GC503-3 of FLEX 10k series. The sampling rate of final implement can operate at the high speed of 400k HZ.
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Huang, Tsung-Wen, and 黃宗文. "The Application of Adaptive Noise Canceller for Hearing Aids." Thesis, 2001. http://ndltd.ncl.edu.tw/handle/56356735592487423206.
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碩士國立成功大學
電機工程學系
89
The structure of traditional hearing aids is relatively simple. They amplify ambient sounds which include speech signal as well as noise. Because noise and human speech signal are amplified at the same time, hearing aid users can’t clearly hear speech signal in noisy environment. By combining adaptive noise canceller and adaptive beamformer, the direction of sound can be used to discriminate speech signal from noise. We have developed a noise cancellation system that based on Griffiths-Jim beamformer. This technique combines NLMS and FLS adaptive algorithms. A real time system was implemented with a TMS320C5402 digital signal processor and some other analog circuits. The hearing aids system was tested by simulated noises. The test results showed that the hearing aids system has directionality. It can reduce noise and improve the Signal-to-Noise Ratio (SNR). 英文摘要——————————————————————————— II 誌謝————————————————————————————— III 目錄————————————————————————————— IV 第一章 緒論————————————————————————— 1 1—1 研究動機——————————————————————— 1 1—2 研究目的——————————————————————— 1 第二章 適應性噪音消除器之原理———————————————— 2 2—1 適應性噪音消除器——————————————————— 2 2—2 NLMS適應性濾波器演算法——————————————— 2 2—3 RLS適應性濾波器演算法———————————————— 3 2—4 FLS適應性濾波器演算法———————————————— 4 2—5 陣列信號處理————————————————————— 6 2—6 Griffiths-Jim波束構成器———————————————— 7 2—7 Griffiths-Jim波束構成器應用於助聽器—————————— 9 第三章 系統設計—————————————————————— 10 3—1 系統架構—————————————————————— 10 3—2 麥克風與放大器——————————————————— 11 3—3 低通濾波器與Anti-aliasing 濾波器——————————— 11 3—4 A/D & D/A轉換器—————————————————— 12 3—5 電壓準位轉換界面—————————————————— 14 3—6 混音器——————————————————————— 14 第四章 軟體設計—————————————————————— 15 4—1 NLMS演算法之適應性波束構成器——————————— 15 4—2 FLS演算法之適應性波束構成器———————————— 16 4—3 定點數與浮點數的差異———————————————— 16 4—4 FLS合併NLMS之適應性波束構成器—————————— 17 第五章 結果與討論————————————————————— 19 5—1 NLMS演算法之適應性波束構成器的實現與測試————— 20 5—2 FLS演算法之適應性波束構成器的實現與測試—————— 27 5—3 模擬語音訊號源以及模擬噪音源測試結果———————— 33 5—4 模擬語音訊號源以及實際噪音源測試結果———————— 35 5—5 討論———————————————————————— 38 5—6 未來展望—————————————————————— 39
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Liao, Hui-Hsiung, and 廖輝雄. "Applied Variable Step Size Algorithm to Dual-Adaptive Noise Canceller." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/87802417601173925190.
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碩士龍華科技大學
電子系碩士班
94
The adaptive filter is an important tool in digital signal processing. Depending on the changing of system conditions, an adaptive filter can be designed to estimate the parameters of the system and to adjust accordingly. Because of this advantage and its simplicity, the time-domain adaptive LMS algorithm has attracted a lot of attention and become one of the most widely used techniques in the past decade. However, because its converging speed is slow and there is no generally applicable rule in deciding the step size in each iteration, this technique is not very easy to implement. For this reason, many researchers have devoted their efforts in solving these problems. This paper derives an algorithm to estimate the step size of the adaptive filter and simulate the behavior of the adaptive noise canceller in real time. The proposed ANC has two adaptive filters: a main filter (MF) and a subfilter (SF). The signal-to-noise ratio (SNR) of input signals is estimated using the SF. To reduce signal distortion in the output signal of the ANC, a step size for coefficient update in the MF is controlled according to the stimated SNR. Moreover, comparative studies on numerical experiments demonstrate the effectiveness of the proposed algorithm.
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Lin, Chun Fan, and 林俊帆. "Study of Adaptive Nonlinear Echo Canceller Using Layered Bilinear Structure." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/11342798593818400962.
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碩士國立交通大學
電子研究所
84
A nonlinear echo canceller, based on the adaptive two-layer bilinear filter structure, is presented. The computation coeifficients are reduced from a general adaptive layered bilinear filter. Computer simulation results show that our methods can improve the echo cancellation performanceefficiently. The ERLE(echo return loss enhancement) over the adaptive linearecho canceller is 15 dB approximately at SNR=30 dB.
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Lee, Yinman, and 李彥文. "Adaptive Decision Feedback Generalized Sidelobe Canceller: Performance Analysis and Applications." Thesis, 2006. http://ndltd.ncl.edu.tw/handle/18174633789388170975.
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博士國立交通大學
電信工程系所
94
The adaptive generalized sidelobe canceller (GSC) is a commonly used device for interference cancellation in array beamforming. However, due to its inherent structure, its convergence is slow. Also, it is sensitive to model mismatch. When model mismatch exists, a phenomenon called signal cancellation will occur, and the performance of the adaptive GSC can be seriously affected. These problems limit the use of adaptive GSC in time-variant systems and also complicate real-world implementations. In this dissertation, we propose a new scheme that can effectively solve these problems. The main idea is to introduce a single-tap decision-directed equalizer and a single-tap feedback filter in the GSC structure, resulting in a decision feedback GSC (DFGSC). The least-mean-square (LMS) algorithm is used for adaptation and the convergence behavior of the adaptive DFGSC is fully analyzed. It is shown that the convergence rate can be greatly enhanced and signal cancellation can be completely avoided. In wireless communications, co-channel interference (CCI) and inter-symbol interference (ISI) are two main factors limiting the system performance. Conventionally, a beamformer is used to reduce CCI while an equalizer is used to compensate ISI. These two devices can be combined into one named space-time equalizer (STE). A training sequence is usually required to train the STE prior to its use. In some applications, however, spatial information corresponding to a desired user is available, but the training sequence is not. Extending the DFGSC approach, we then propose an adaptive decision feedback STE to cope with this problem. Our scheme consists of an adaptive DFGSC, a blind decision feedback equalizer (DFE), and a channel estimator. With a specially designed structure, the proposed blind DFE, aided by the estimated channel, can better resist error propagation effect inherent in a DFE. As previously, adaptation operations are implemented with the LMS algorithm and convergence analysis is given as well. Simulations show that the proposed adaptive decision feedback STE is effective in mitigating both CCI and ISI even in severe channel environments. The channel-aided blind DFE structure can be further applied to the general adaptive minimum mean-squared-error DFE (MMSE-DFE), called the adaptive channel-aided DFE (ACA-DFE). We also demonstrate that the ACA-DFE can be extended to multiple-input multiple-out (MIMO) systems and it can outperform the conventional adaptive MIMO MMSE-DFE. Adaptive parallel interference cancellation (PIC) has been recently introduced in the signal detection of MIMO systems. Conventional PIC uses the MMSE criterion for parameter adaptation. However, it is known that the MMSE criterion cannot achieve the minimum bit-error-rate (MBER). Also, it suffers from the error propagation problem when operated in time-variant channels. In the last part of the dissertation, an adaptive two-stage PIC detection scheme with the minimum variance (MV) criterion is proposed to solve the problems. Adaptation with the MV criterion is then realized with the DFGSC. In the first-stage cancellation, a dual-DFGSC configuration being effective in time-variant channel environments is developed. Due to the good performance of the first-stage processing, only matched filtering is required in the second stage to achieve near optimum results. The LMS algorithm is employed and its convergence behavior is also examined. Simulation results show that the proposed two-stage DFGSC-PIC detection significantly outperforms the conventional MMSE-PIC detection.
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Liao, Wei-Shun, and 廖偉舜. "Pulmonary Recording System Using Transform Domain Adaptive Ambient Noise Canceller." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/47440777299792938189.
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碩士國立臺灣大學
電機工程學研究所
90
The recording process of pulmonary sounds is the first and very important step of the researches about pulmonary sounds. The pulmonary sound signal would corrupted by the ambient noise during and the post-process of the pulmonary sound signal would be erroneous because of the ambient noise interference. Therefore, the aim of this research is to develop a pulmonary recording system that could remove the ambient noise recorded in pulmonary sounds effectively. The solution this thesis proposed is to use the structure of transform domain adaptive filter (TDAF) with Walsh-Hadamard transform (WHT) to replace the structure of traditional adaptive filter. This structure would improve the convergence speed of adaptive filter algorithm so that the ambient noise canceller in pulmonary recording system would effectively remove the noise with non-stationary property result from the time-varying property of outer environment. This thesis also proposes some hardware improvements. First, the analog circuit has been modified so that it is smaller in dimension and needs lower power consumption. Second, the digital signal processor (DSP) has been used for central control unit of this system and single power input architecture for power supply circuit. Therefore, we could use this system for long-term and real-time expert system that could monitor individual health condition everywhere in the future.
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WU, XIAN-HUANG, and 吳先晃. "Adaptive echo canceller for 2-wire full-duplex voiceband data transmission." Thesis, 1986. http://ndltd.ncl.edu.tw/handle/62938755851587401051.
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Friedman, Richard M. "A combined channel-modified adaptive array MMSE canceller and viterbi equalizer." Thesis, 2003. http://library1.njit.edu/etd/fromwebvoyage.cfm?id=njit-etd2003-085.
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黃志友. "Pre-rake transmit diversity with adaptive interference canceller for CDMA systems." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/42948435169104633091.
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Liao, Ching-Wei, and 廖敬瑋. "Robust Adaptive Beamforming Using Modified Generalized Sidelobe Canceller Under Non-ideal Environments." Thesis, 2015. http://ndltd.ncl.edu.tw/handle/12687364340673925278.
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碩士國立臺灣大學
電信工程學研究所
103
LCMV (Linearly Constrained Minimum Variance) beamformer is one of the important and basic techniques in the antenna array signal processing. It aims to find the optimal solution for the weight vector by maintaining the desired array response while minimizing the array output power. Thus, the beamformer has a great ability to suppress interference and noise. The GSC (Generalized Sidelobe Canceller) derived from LCMV beamformer provides a simple way for implementing and makes the implementation of the LCMV much more efficient. However, the performance of the aforementioned beamformers is known to be degraded dramatically in the presence of steering vector error because of the signal cancellation phenomenon which leads the desired signal to be suppressed as interference. To improve the robustness, we propose a method called MGSC (modified GSC) evolved from the GSC. The MGSC uses a novel block matrix which can not only avoids the signal cancellation phenomenon but also decreases the computational complexity by diminishing the size of the adaptive weight vector. We also give the MGSC an ability to estimate steering vector and it only needs few computations. Then, we propose a way to make the MGSC combined with the existing algorithms. It can also improve the performance and decrease the computational complexity at the same time. Finally, we combine spatial smoothing technique with the MGSC to against coherent interference and steering vector error simultaneously.
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Lin, Po-Yi, and 林柏怡. "Robust Adaptive Beamforming using A New Generalized Sidelobe Canceller Under Non-ideal Environment." Thesis, 2016. http://ndltd.ncl.edu.tw/handle/94293961173296280648.
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碩士國立臺灣大學
電信工程學研究所
104
Adaptive array beamforming is the technique which can not only extracts signals of interest from specific angles but also suppresses interferences and noise. LCMV(Linearly Constrained Minimum Variance) beamformer and GSC(Generalized Sidelobe Canceller) are the important techniques in the antenna array signal processing which need the direction of the desired signal as a priori information. The main part of this thesis is to modify GSC(Generalized Sidelobe Canceller) and then proposes the NBMGSC (Noise Block Modified Generalized Sidelobe Canceller) by combining noise block matrix and MGSC(Modified Generalized Sidelobe Canceller). This approach can not only decreases the computational complexity but also suppresses the interference to improve the system performance. We also let NBMGSC(Noise Block Modified Generalized Sidelobe Canceller) combine with the proposed robustness algorithms. It can decrease the computational complexity and improve the system performance to against the multiple non-ideal environments. For the non-ideal environments, it includes steering vector mismatch, coherent environment, mutual coupling effect and unknown mutual coupling and then we combine the above non-ideal environments and we use the proposed method to against the multiple non-ideal environments.
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Wei, Fan Yu, and 范育維. "Multitone Modulation Systemswith Interference Canceller or Adaptive Number of Tones in Wireless Channels." Thesis, 2002. http://ndltd.ncl.edu.tw/handle/43529215994034312663.
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碩士國立臺灣科技大學
電子工程系
90
Multi-media communication and high-speed data transmission are parts of the main goals for the development of the third generation wireless communication system. The multi-path fading of the wireless channels would cause severe inter-symbol interference (ISI) for high-speed data transmission and therefore reduce the performance of the communication system. To overcome the ISI problem for the high-speed data transmission, complex channel equalization at receivers is required. An alternative way to overcome the problem is to use multi-carrier modulation scheme. In this thesis, we apply multi-tone modulation scheme based on inverse discrete Fourier transform (IDFT) to transmit high-speed data. In the receiver, we use the matched filter (MF) bank or RAKE receiver bank to detect the transmitted signal. However, the multi-path fading channel has a non-flat impulse transfer function. Therefore, the sub-carriers of the MT modulation system are no longer orthogonal at the receiver and cause inter-carrier interference (ICI). To solve the problem, interference canceller is required at the receiver. Besides the conventional interference canceller, which is a decision feedback equalizer (DFE), we propose an eigendecomposition-derived algorithm to eliminate the ISI and the ICI. Moreover, closed-form equations for the computation of bit error rate (BER) in the proposed MT systems are derived. We also propose MT modulation system that has adaptive number of tones. Then, the adaptive number of tones is computed to minimize the BER, given the channel characteristics and the available signal to noise ratio. Two cases, where channel parameters can be exactly estimated or statistically estimated, respectively, are investigated. It is observed in simulation that the RAKE reception is better than the MF reception in BER performance. Simulation also shows that the MT modulation system using adaptive number of tones outperforms the conventional MT modulation system.
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Chen, CHI-HE, and 陳麒合. "Fetal ECG Extraction and Analysis from Composite Maternal ECG using an Adaptive Noise Canceller." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/66112000204508241649.
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碩士亞洲大學
生物資訊學系碩士班
99
Electorcardiography (ECG) is one of the best ways to measure and diagnose abnormal rhythms of the heart. Its noninvasive nature is especially beneficial for diagnosis of fetal heart defects in advance of delivery. However, there is no appropriate method for noninvasively measuring the fetal ECGs. In this study, we propose a method to cancel noise and drifting and further to extract fetal ECG effectively by an adaptive filter from maternal abdomen and thorax ECGs. In addition, an integrated analysis software, Fetal ECG Analysis Environment (FEAE), for fetal ECG signal extraction and evaluation using the proposed method is designed and implemented in MATLAB GUIDE environment for future study.
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Yang, Yun-Si, and 楊盷羲. "Porting of FreeRTOS with real-time Adaptive Noise Canceller and Bluetooth Transmission on ARM Cortex M0." Thesis, 2013. http://ndltd.ncl.edu.tw/handle/87870614214021948049.
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碩士東南科技大學
電機工程研究所
101
Considering the rapid arrival of aging population society in Taiwan, the prevalence of the Android platform, and the competitive price of the 32 bits Cortex-M microcontroller by ARM corp., the long-term home-care medical instruments combined with cloud medical services become more and more important in the future. This study adopts a cheap Cortex-M0 microcontroller ($0.5pcs or so for asking price) and uses its embedded ADC to convert two analog input signals, then the digital signals are processed by adaptive noise canceller (ANC). The experiments are arranged to use different step size to observe the filtering effect of interference and noise by different frequency and amplitude in order to discover the stable system structure for mass manufacture of individual mobile medical instruments. The last part of this study is transferred the output of ANC to Android terminal by bluetooth protocol. The whole system realizes a individual mobile acquisition and transmission system for biomedical signals.
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Jiun, Yang Yu, and 楊渝軍. "Combination of two local adaptive antenna arrayand an array combiner for cochannel interference canceller in DS-SSMA." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/33216798547038398383.
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碩士國立臺灣科技大學
電子工程學系
84
In the realization of code division multiple access based on a spread-spectrum communication system, i.e., spread spec- trum multiple access (SSMA), reduction of cochannel inter- ference is important problem. This paper proposes an adaptive array system which can improve performance by combining two local adaptive antenna arrays and an array output combiner. While one of two local antenna arrays cannot suppress inter- ference sources with arrival angles the same as desired user's, the other local array can reject them. The proposed system can achieve stable acquisition and low error rate of demodulated data even in a heavy interference channel where a conventional array antenna system cannot achieve satisfac- tory acquisition.
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Yang, Yu-Jun, and 楊渝軍. "Combination of two Local Adaptive Antenna Arrays and An Array Output Combiner for Cochannel Interference Canceller in DS-SSMA System." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/14527691585201697416.
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