本文選題：多徑衰落 + 自適應(yīng)均衡　； 參考：《中北大學(xué)》2017年碩士論文

【摘要】：有效性和可靠性是設(shè)計通信系統(tǒng)時不得不考慮的兩個問題,在無線通信系統(tǒng)中,多徑效應(yīng)和多普勒頻移是造成碼間干擾,導(dǎo)致頻率和時間選擇性衰落的主要原因。自適應(yīng)均衡技術(shù)應(yīng)運而生,由于自適應(yīng)均衡器可以產(chǎn)生與信道特性相反的特性,用來消除信道的的時變多徑傳播特性帶來的干擾,在現(xiàn)代數(shù)字微波通信系統(tǒng)中得到了廣泛的應(yīng)用。本文主要針對數(shù)字微波通信系統(tǒng)中信道的時變多徑衰落的特性,對自適應(yīng)均衡器及其算法進行了研究,并在TI公司生產(chǎn)TMS320VC5509A定點DSP開發(fā)板上進行了實現(xiàn)。首先本文介紹了無線通信中的信道以及所用的模型,并給出了信道的等效離散時間模型。緊接著對自適應(yīng)均衡器的結(jié)構(gòu)分類以及工作原理做了闡述,對自適應(yīng)均衡算法及其調(diào)節(jié)準(zhǔn)則做了理論性的介紹分析,并在MATLAB軟件環(huán)境下模擬實際環(huán)境,對使用LMS和RLS算法實現(xiàn)自適應(yīng)均衡器的性能進行了驗證分析。隨后在VC6.0環(huán)境下對自適應(yīng)均衡器進行了模擬仿真,用C++語言編寫了矩陣類,實現(xiàn)了矩陣的基本操作,產(chǎn)生了高斯白噪聲和數(shù)據(jù)輸入矩陣,在此基礎(chǔ)上實現(xiàn)了RLS和LMS兩種算法的自適應(yīng)均衡器,為利用DSP實現(xiàn)自適應(yīng)均衡器打下基礎(chǔ)。最后,先通過建立自適應(yīng)噪聲消除模型,在TMS320VC5509A定點DSP開發(fā)板上對使用LMS和RLS兩種算法自適應(yīng)均衡器中濾波部分的功能進行了仿真驗證,達到了理想要求,最終,在上述開發(fā)板上實現(xiàn)了采用LMS算法的自適應(yīng)均衡器,并對不同信道不同信噪比不同步長以及信道發(fā)生變化的時變信道的實驗結(jié)果進行了比較和分析。本系統(tǒng)主要采用C語言編寫完成,具有很好的可讀性和可移植性,維護方便,參數(shù)設(shè)置靈活,具有很好的應(yīng)用前景。
[Abstract]:Effectiveness and reliability are two problems that must be considered when designing communication systems. In wireless communication systems, multipath effect and Doppler frequency shift are the main causes of inter-symbol interference and frequency and time selective fading. The adaptive equalization technique emerges as the times require. Because the adaptive equalizer can produce the opposite characteristics of the channel, it is used to eliminate the interference caused by the time-varying multipath propagation of the channel. It is widely used in modern digital microwave communication system. In this paper, the adaptive equalizer and its algorithm are studied in view of the time-varying multipath fading characteristics of the channel in digital microwave communication system, and implemented on the TMS320VC5509A fixed-point DSP development board produced by TI Company. Firstly, this paper introduces the channel and the model used in wireless communication, and gives the equivalent discrete time model of the channel. Then the structure classification and working principle of the adaptive equalizer are expounded, and the adaptive equalization algorithm and its adjustment criterion are introduced and analyzed theoretically, and the actual environment is simulated in the environment of MATLAB software. The performance of adaptive equalizer using LMS and RLS algorithms is verified and analyzed. Then the adaptive equalizer is simulated under the VC6.0 environment, and the matrix class is written in C language, which realizes the basic operation of the matrix and produces Gao Si white noise and data input matrix. On this basis, the adaptive equalizer of RLS and LMS algorithms is implemented, which lays the foundation for the realization of adaptive equalizer using DSP. Finally, by establishing an adaptive noise cancellation model, the functions of filtering part of adaptive equalizer using LMS and RLS algorithms are simulated on the TMS320VC5509A fixed-point DSP development board. An adaptive equalizer using LMS algorithm is implemented on the above development board, and the experimental results of time-varying channels with different signal-to-noise ratio (SNR) nonsynchronous length and channel variation are compared and analyzed. The system is mainly written in C language, which has good readability and portability, convenient maintenance, flexible parameter setting, and has a good application prospect.
【學(xué)位授予單位】：中北大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2017
【分類號】：TN715

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