本文選題：自適應(yīng)濾波 + 噪聲抵消��； 參考：《河北科技大學(xué)》2015年碩士論文

【摘要】：本文首先研究了自適應(yīng)濾波器的基本原理、系統(tǒng)組成結(jié)構(gòu)以及實(shí)際應(yīng)用,深入分析了以最小均方誤差(MMSE)為準(zhǔn)則的LMS算法和基于最小二乘準(zhǔn)則的RLS算法;分析了兩種算法的優(yōu)點(diǎn)以及缺點(diǎn),并進(jìn)行了對比;對于傳統(tǒng)的LMS算法的收斂速度以及穩(wěn)態(tài)誤差之間的無法兼顧,在誤差接近零時出現(xiàn)較大幅度的波動以及在低信噪比情況下性能降低等問題,提出了一種變步長LMS算法,通過仿真驗(yàn)證本文提出的算法優(yōu)越于傳統(tǒng)的濾波算法以及改進(jìn)的算法。其次在自適應(yīng)濾波算法的基礎(chǔ)上,對于消噪系統(tǒng)如何使用自適應(yīng)算法消除噪聲進(jìn)行系統(tǒng)的分析,自適應(yīng)噪聲抵消系統(tǒng)的實(shí)質(zhì)就是運(yùn)用自適應(yīng)濾波理論有效的消除噪聲,對于傳統(tǒng)消噪算法,重點(diǎn)分析了結(jié)構(gòu)、性能以及衡量指標(biāo)。對于傳統(tǒng)的自適應(yīng)噪聲抵消算法難以解決脈沖噪聲以及參考通道信號的難以獲取的問題,提出了一種改進(jìn)的自適應(yīng)噪聲抵消算法,并與傳統(tǒng)的算法進(jìn)行了分析以及對比。仿真結(jié)果證明對于周期性信號提取問題本文提出的算法具有一定的優(yōu)越性并且克服了傳統(tǒng)算法處于脈沖噪聲干擾的環(huán)境下性能惡化的問題。最后,基于對自適應(yīng)噪聲技術(shù)理論分析以及研究的基礎(chǔ)上,運(yùn)用TI公司的高速處理器件TMS320VC5509A、性能優(yōu)越的音頻處理芯片TLV320AIC23及其它相關(guān)模塊共同構(gòu)成一個自適應(yīng)噪聲抵消系統(tǒng),利用DSP集成開發(fā)環(huán)境(CCS4.2)的軟件仿真模式下,實(shí)現(xiàn)了傳統(tǒng)的LMS和RLS及本文提出的自適應(yīng)濾波算法。在硬件仿真器模式下,分別選取noise92庫里的白噪聲和工廠背景噪聲作為環(huán)境噪聲進(jìn)行語音信號消噪,采用簡單而且容易實(shí)現(xiàn)的LMS算法作為核心濾波算法,選取硬件仿真器進(jìn)行下載鏈接,在此系統(tǒng)平臺上進(jìn)行語音信號實(shí)時消噪處理。本文詳細(xì)介紹了噪聲抵消系統(tǒng)結(jié)構(gòu)組成部分,并給出了DSP以及音頻芯片AIC23的鏈接框圖和程序流程圖,最后給出了測試結(jié)果,具有一定的參考價值。
[Abstract]:In this paper, the basic principle, system structure and practical application of adaptive filter are studied. The LMS algorithm based on minimum mean square error (MMSE) and the RLS algorithm based on least square criterion are deeply analyzed. The advantages and disadvantages of the two algorithms are analyzed and compared, and the convergence speed and steady-state error of the traditional LMS algorithm can not be taken into account. In this paper, a variable step size LMS algorithm is proposed, which is superior to the traditional filtering algorithm and the improved algorithm by simulation, due to the large fluctuation of the error near 00:00 and the degradation of the performance in the case of low signal-to-noise ratio (SNR). Secondly, on the basis of adaptive filtering algorithm, how to use adaptive algorithm to eliminate noise is analyzed systematically. The essence of adaptive noise cancellation system is to use adaptive filtering theory to eliminate noise effectively. For the traditional denoising algorithm, the structure, performance and measurement index are analyzed. For the traditional adaptive noise cancellation algorithm is difficult to solve the problem of impulse noise and reference channel signal difficult to obtain, an improved adaptive noise cancellation algorithm is proposed, and compared with the traditional algorithm. The simulation results show that the proposed algorithm is superior to the periodic signal extraction problem and overcomes the problem that the traditional algorithm is in the environment of impulse noise interference. Finally, based on the theoretical analysis and research of adaptive noise technology, an adaptive noise cancellation system is composed of TMS320VC5509A, TLV320AIC23, an audio processing chip with superior performance, and other related modules. Under the software simulation mode of DSP integrated development environment (CCS 4.2), the traditional LMS and RLS and the adaptive filtering algorithm proposed in this paper are realized. In the hardware simulator mode, the white noise and factory background noise in noise92 library are selected as the ambient noise to remove the noise of speech signal, and the simple and easy to realize the algorithm is used as the core filtering algorithm. The hardware simulator is selected to download the link, and the real-time speech signal denoising processing is carried out on this system platform. In this paper, the structure of noise cancellation system is introduced in detail, and the link block diagram and program flow chart of DSP and AIC23 are given. Finally, the test results are given, which have certain reference value.
【學(xué)位授予單位】：河北科技大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2015
【分類號】：TN911.4;TN713

【參考文獻(xiàn)】

相關(guān)期刊論文 前1條

1 呂春英;敖偉;張洪順;;一種新的變步長LMS算法[J];通信技術(shù);2011年03期

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本文編號：2014647