子帶自適應(yīng)濾波技術(shù)研究及其應(yīng)用
發(fā)布時(shí)間：2018-12-10 07:47
【摘要】：自適應(yīng)濾波器在通信、聲納、雷達(dá)、音頻處理、圖像處理等領(lǐng)域有著廣泛的應(yīng)用,是自適應(yīng)信號處理中最受關(guān)注的研究課題之一,其性能主要由所使用的自適應(yīng)算法和濾波器結(jié)構(gòu)決定。子帶自適應(yīng)濾波器具有特殊的結(jié)構(gòu),能夠?qū)斎胄盘栠M(jìn)行頻帶分割與抽取,從而減小輸入信號的相關(guān)性、提高收斂速度、降低對信號處理速度的要求,近年來被廣泛應(yīng)用于信號處理領(lǐng)域。本文依據(jù)課題需要,針對子帶自適應(yīng)濾波器中的濾波器組設(shè)計(jì)和自適應(yīng)濾波算法展開深入研究,主要內(nèi)容及創(chuàng)新點(diǎn)概況如下:1.研究了均勻子帶濾波器組的基本結(jié)構(gòu)以及原型濾波器的設(shè)計(jì)方法。針對均勻子帶余弦調(diào)制濾波器組中,采用間接設(shè)計(jì)法得到的原型濾波器恢復(fù)出的信號幅度存在失真較大的問題,首先依據(jù)重構(gòu)條件和過渡帶的平方根余弦滾降特性設(shè)計(jì)代價(jià)函數(shù),然后通過選擇合適的原型濾波器通帶截止頻率,使其3dB通帶截止頻率接近理想濾波器的通帶截止頻率,最后通過迭代的方式最小化代價(jià)函數(shù)并獲得最佳原型濾波器。仿真實(shí)驗(yàn)表明,相比于同類設(shè)計(jì)方法,改進(jìn)后的原型濾波器具有更好的重構(gòu)性能。2.針對系統(tǒng)辨識中的應(yīng)用,設(shè)計(jì)了兩種變參數(shù)子帶自適應(yīng)濾波器。針對仿射投影歸一化子帶自適應(yīng)濾波算法收斂速度和穩(wěn)態(tài)失調(diào)存在的折中問題,設(shè)計(jì)了使均方偏差上界最小化的變步長矩陣替代固定步長,該變步長矩陣由各子帶的步長參數(shù)構(gòu)成,相對原有算法僅有較低的復(fù)雜度增加。實(shí)驗(yàn)仿真表明,該方法相比變參數(shù)歸一化子帶自適應(yīng)濾波類算法具有更好的收斂性能、跟蹤性能以及對脈沖干擾的魯棒性。針對仿射投影符號子帶自適應(yīng)濾波算法收斂速度和穩(wěn)態(tài)失調(diào)存在矛盾的問題,提出了一種變正則化參數(shù)設(shè)計(jì)方法。該方法采用隨機(jī)梯度下降法來更新正則化參數(shù)從而使得均方偏差最小化,仿真驗(yàn)證了其收斂性能和對脈沖干擾以及double-talk的魯棒性好于變參數(shù)符號子帶自適應(yīng)濾波類算法。3.針對信道均衡中的應(yīng)用,設(shè)計(jì)了兩種變參數(shù)子帶均衡器。通過分析發(fā)現(xiàn)多帶結(jié)構(gòu)可以有效的克服傳統(tǒng)子帶均衡器輸出端存在混疊分量的問題,在此基礎(chǔ)上提出了兩種基于多帶結(jié)構(gòu)的變參數(shù)子帶均衡器,來解決定參數(shù)均衡器收斂速度和穩(wěn)態(tài)失調(diào)之間的矛盾。第一種是在符號子帶均衡算法的基礎(chǔ)上,將每次迭代中最小化后驗(yàn)誤差向量的能量作為代價(jià)函數(shù),來設(shè)計(jì)變步長矩陣,并用于下一次迭代;第二種是在歸一化子帶均衡算法的基礎(chǔ)上,將最小化誤差向量的能量作為代價(jià)函數(shù),采用隨機(jī)梯度下降法來更新正則化參數(shù),從而得到該均衡算法的變正則化參數(shù)。兩種方法都能較好的解決子帶均衡器的快收斂速度和低穩(wěn)態(tài)失調(diào)之間的矛盾,從而獲得良好的整體性能。
[Abstract]:Adaptive filter is widely used in communication, sonar, radar, audio processing, image processing and so on. It is one of the most concerned research topics in adaptive signal processing. Its performance is mainly determined by the adaptive algorithm and filter structure used. The sub-band adaptive filter has a special structure, it can segment and decimate the input signal in the frequency band, thus reducing the correlation of the input signal, improving the convergence speed and reducing the requirement of the signal processing speed. In recent years, it has been widely used in the field of signal processing. In this paper, the design of filter banks and adaptive filtering algorithms in subband adaptive filters are studied. The main contents and innovations are as follows: 1. The basic structure of uniform subband filter bank and the design method of prototype filter are studied. In the uniform subband cosine modulation filter bank, the amplitude of the recovered signal from the prototype filter obtained by indirect design method has the problem of large distortion. First, the cost function is designed according to the reconstruction condition and the square root cosine roll down characteristic of the transition band, and then the 3dB passband cut-off frequency is close to that of the ideal filter by selecting the appropriate passband cut-off frequency of the prototype filter. Finally, the cost function is minimized and the optimal prototype filter is obtained by iterative method. The simulation results show that the improved prototype filter has better reconstruction performance than the similar design method. 2. Aiming at the application of system identification, two kinds of variable parameter subband adaptive filters are designed. Aiming at the convergence rate of affine projection normalized subband adaptive filtering algorithm and the trade-off between steady state misalignment, a variable step matrix is designed to minimize the upper bound of mean square deviation instead of fixed step size. The variable step size matrix is composed of the step size parameters of each subband, and the complexity of the variable step matrix is only lower than that of the original algorithm. Experimental results show that the proposed method has better convergence performance, tracking performance and robustness to impulse interference than the variable parameter normalized subband adaptive filtering algorithm. Aiming at the contradiction between convergence speed and steady state misalignment of affine projection symbol subband adaptive filtering algorithm, a variable regularization parameter design method is proposed. The stochastic gradient descent method is used to update the regularization parameters to minimize the mean square deviation. Simulation results show that its convergence performance and robustness to pulse interference and double-talk are better than those of variable parameter symbol subband adaptive filtering algorithms. Aiming at the application of channel equalization, two kinds of variable parameter subband equalizers are designed. It is found that multi-band structure can effectively overcome the problem of aliasing component in the output of traditional sub-band equalizer. Based on this, two kinds of variable parameter sub-band equalizers based on multi-band structure are proposed. To solve the contradiction between convergence rate and steady state misalignment of constant parameter equalizer. The first is to design the variable step size matrix by minimizing the energy of the posteriori error vector as a cost function on the basis of the symbolic subband equalization algorithm and using it for the next iteration. The second is based on the normalized subband equalization algorithm. The energy of the minimized error vector is taken as the cost function, and the regularization parameters are updated by the stochastic gradient descent method, and the variable regularization parameters of the equalization algorithm are obtained. Both methods can solve the contradiction between the fast convergence speed of sub-band equalizer and the low steady-state misalignment, and obtain good overall performance.
【學(xué)位授予單位】：解放軍信息工程大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2015
【分類號】：TN713
本文編號：2370230
[Abstract]:Adaptive filter is widely used in communication, sonar, radar, audio processing, image processing and so on. It is one of the most concerned research topics in adaptive signal processing. Its performance is mainly determined by the adaptive algorithm and filter structure used. The sub-band adaptive filter has a special structure, it can segment and decimate the input signal in the frequency band, thus reducing the correlation of the input signal, improving the convergence speed and reducing the requirement of the signal processing speed. In recent years, it has been widely used in the field of signal processing. In this paper, the design of filter banks and adaptive filtering algorithms in subband adaptive filters are studied. The main contents and innovations are as follows: 1. The basic structure of uniform subband filter bank and the design method of prototype filter are studied. In the uniform subband cosine modulation filter bank, the amplitude of the recovered signal from the prototype filter obtained by indirect design method has the problem of large distortion. First, the cost function is designed according to the reconstruction condition and the square root cosine roll down characteristic of the transition band, and then the 3dB passband cut-off frequency is close to that of the ideal filter by selecting the appropriate passband cut-off frequency of the prototype filter. Finally, the cost function is minimized and the optimal prototype filter is obtained by iterative method. The simulation results show that the improved prototype filter has better reconstruction performance than the similar design method. 2. Aiming at the application of system identification, two kinds of variable parameter subband adaptive filters are designed. Aiming at the convergence rate of affine projection normalized subband adaptive filtering algorithm and the trade-off between steady state misalignment, a variable step matrix is designed to minimize the upper bound of mean square deviation instead of fixed step size. The variable step size matrix is composed of the step size parameters of each subband, and the complexity of the variable step matrix is only lower than that of the original algorithm. Experimental results show that the proposed method has better convergence performance, tracking performance and robustness to impulse interference than the variable parameter normalized subband adaptive filtering algorithm. Aiming at the contradiction between convergence speed and steady state misalignment of affine projection symbol subband adaptive filtering algorithm, a variable regularization parameter design method is proposed. The stochastic gradient descent method is used to update the regularization parameters to minimize the mean square deviation. Simulation results show that its convergence performance and robustness to pulse interference and double-talk are better than those of variable parameter symbol subband adaptive filtering algorithms. Aiming at the application of channel equalization, two kinds of variable parameter subband equalizers are designed. It is found that multi-band structure can effectively overcome the problem of aliasing component in the output of traditional sub-band equalizer. Based on this, two kinds of variable parameter sub-band equalizers based on multi-band structure are proposed. To solve the contradiction between convergence rate and steady state misalignment of constant parameter equalizer. The first is to design the variable step size matrix by minimizing the energy of the posteriori error vector as a cost function on the basis of the symbolic subband equalization algorithm and using it for the next iteration. The second is based on the normalized subband equalization algorithm. The energy of the minimized error vector is taken as the cost function, and the regularization parameters are updated by the stochastic gradient descent method, and the variable regularization parameters of the equalization algorithm are obtained. Both methods can solve the contradiction between the fast convergence speed of sub-band equalizer and the low steady-state misalignment, and obtain good overall performance.
【學(xué)位授予單位】：解放軍信息工程大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2015
【分類號】：TN713
【參考文獻(xiàn)】
相關(guān)期刊論文 前8條
1 趙艷;羅勝恩;萬堅(jiān);;余弦調(diào)制子帶濾波器組迭代設(shè)計(jì)的改進(jìn)算法[J];數(shù)據(jù)采集與處理;2013年01期
2 張?zhí)祢U;譚方青;高春霞;黃烈超;;一種新的窗函數(shù)法設(shè)計(jì)余弦調(diào)制濾波器組系統(tǒng)[J];系統(tǒng)工程與電子技術(shù);2011年12期
3 蔣俊正;水鵬朗;周芳;;M帶余弦調(diào)制濾波器組的迭代設(shè)計(jì)算法[J];電路與系統(tǒng)學(xué)報(bào);2011年06期
4 趙艷;萬堅(jiān);;一種單載波寬帶信號非線性均衡技術(shù)[J];計(jì)算機(jī)工程與應(yīng)用;2012年05期
5 張煒;戴旭初;許小東;;基于非均勻子帶分解的寬帶線性盲均衡器[J];電子學(xué)報(bào);2010年04期
6 張煒;戴旭初;;一種分?jǐn)?shù)間隔恒模算法的寬帶盲均衡器[J];中國科學(xué)技術(shù)大學(xué)學(xué)報(bào);2009年09期
7 張煒;金光明;戴旭初;;一種基于子帶分解技術(shù)的寬帶盲均衡器結(jié)構(gòu)和算法[J];電子與信息學(xué)報(bào);2009年03期
8 張子敬,焦李成;M帶余弦調(diào)制濾波器組的設(shè)計(jì)[J];電子學(xué)報(bào);2001年01期
，本文編號：2370230
本文鏈接：http://sikaile.net/kejilunwen/dianzigongchenglunwen/2370230.html
最近更新
教材專著
