本文選題：回波消除 + 改進(jìn)NLMS��； 參考：《哈爾濱工業(yè)大學(xué)》2015年碩士論文

【摘要】：大多數(shù)聽力受損患者依靠佩戴數(shù)字助聽器補(bǔ)償聽力,然而市場上各種助聽器良莠不齊,體現(xiàn)性能差異的一個主要方面是助聽器中回波的處理�；夭▽φZ音的清晰度和舒適度都有很大的影響。為了解決助聽器中回波造成的問題,本文針對自適應(yīng)算法在數(shù)字助聽器回波消除中的應(yīng)用進(jìn)行研究和實現(xiàn)。最小均方誤差(Least Mean Square,LMS)算法常常被用于回波消除,然而由于收斂速度慢,誤差較大,不能滿足助聽器系統(tǒng)實時性和清晰度的要求。因此本文采用具有更高性能的歸一化的最小均方誤差(Normalized Least Mean Square,NLMS)算法,而且在傳統(tǒng)的NLMS算法上進(jìn)行改進(jìn),對系數(shù)更新頻率、步長更新方式進(jìn)行調(diào)整,更好地解決了數(shù)字助聽器中回波消除問題。本文中,數(shù)字助聽器回波消除模塊設(shè)計包括:回波檢測算法、改進(jìn)的NLMS算法、回波時延估計算法以及非線性處理(Non-Linear Processing,NLP)算法,以實現(xiàn)語音信號處理的準(zhǔn)確性和實時性。在算法模塊入口,加入回波檢測裝置,當(dāng)回波能量低于閾值時不對輸入語音處理,裝置可以避免濾波器的頻繁跳轉(zhuǎn),保證處理效果的同時節(jié)省系統(tǒng)開銷。對NLMS算法的改進(jìn)包括:分批次更新濾波器系數(shù),同時對步長迭代加入控制因子。因此,改進(jìn)算法具有較小的運(yùn)算量和較快的收斂速度�；夭〞r延估計算法用來估計已處理語音信號與期望信號之間的延時。利用這一延時,兩個緩沖數(shù)組能夠?qū)R以保證輸入數(shù)據(jù)的正確處理。時延估計算法對期望信號和已處理信號進(jìn)行互相關(guān),并將結(jié)果與期望信號自相關(guān)結(jié)果再次進(jìn)行互相關(guān)。延時估計通過兩次互相關(guān)方式對延時進(jìn)行估計,具有更強(qiáng)抗噪能力和穩(wěn)定性。NLP算法利用一個濾波器依據(jù)幅值大小對信號進(jìn)行濾波,阻止低幅值信號并讓高幅值信號通過。通過這種處理,可以實現(xiàn)回波殘留的進(jìn)一步消除,使語音輸出具有更好的舒適度。在MATLAB上對改進(jìn)的NLMS算法進(jìn)行仿真,分析該算法與傳統(tǒng)算法的性能對比。之后利用Cool EditPro軟件對處理后的語音同其他算法處理語音進(jìn)行分析比較。從仿真結(jié)果上看,改進(jìn)算法要強(qiáng)于傳統(tǒng)算法。改進(jìn)算法擁有較快的收斂速度、更小的回波殘留和較好的舒適度。
[Abstract]:Most hearing loss patients rely on digital hearing aids to compensate for their hearing. However, there are different kinds of hearing aids in the market. One of the main aspects of performance difference is the processing of echo in hearing aids. Echo plays an important role in articulation and comfort of speech. In order to solve the problem caused by echo in hearing aid, the application of adaptive algorithm in digital hearing aid echo cancellation is studied and implemented in this paper. Least mean square error (LMS) algorithm is often used for echo cancellation. However, due to the slow convergence rate and large error, it can not meet the requirements of real-time and clarity of hearing aid system. Therefore, the Normalized least mean Square-NLMS (NLMS) algorithm, which has higher performance, is adopted in this paper, and the traditional NLMS algorithm is improved to adjust the updating frequency and step size of the coefficients. The problem of echo cancellation in digital hearing aid is better solved. In this paper, digital hearing aid echo cancellation module design includes: echo detection algorithm, improved NLMS algorithm, echo time delay estimation algorithm and Non-Linear processing (NLP) algorithm to achieve the accuracy and real-time speech signal processing. At the entrance of the algorithm module, the echo detection device is added, when the echo energy is lower than the threshold value, the input voice is not processed correctly, so the device can avoid the frequent jump of the filter and save the system cost while ensuring the processing effect. The improvements to the NLMS algorithm include updating the filter coefficients in batches and adding control factors to step size iterations. Therefore, the improved algorithm has less computation and faster convergence speed. The echo delay estimation algorithm is used to estimate the delay between the processed speech signal and the desired signal. With this delay, the two buffering arrays can be aligned to ensure the correct processing of the input data. The time delay estimation algorithm cross-correlates the desired signal with the processed signal, and the result is cross-correlated with the autocorrelation result of the desired signal. The delay estimation can estimate the delay by two cross-correlation methods. It has stronger anti-noise ability and stability. NLP algorithm uses a filter to filter the signal according to the amplitude to prevent the low-amplitude signal and let the high-amplitude signal pass through. Through this treatment, the echo residue can be further eliminated and the speech output has better comfort. The improved NLMS algorithm is simulated on MATLAB, and its performance is compared with that of the traditional algorithm. Then, Cool EditPro software is used to analyze and compare the processed speech with other algorithms. From the simulation results, the improved algorithm is better than the traditional algorithm. The improved algorithm has faster convergence speed, smaller echo residue and better comfort.
【學(xué)位授予單位】：哈爾濱工業(yè)大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2015
【分類號】：TN912.3

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