發(fā)布時間：2018-06-21 08:11

  本文選題：嘯叫抑制 + 自適應算法　； 參考：《電子科技大學》2014年碩士論文

【摘要】：聲反饋是出現(xiàn)在劇院、多媒體教室、會議室等公共擴聲系統(tǒng)中的常見問題,它經(jīng)常使音頻擴聲系統(tǒng)的性能發(fā)生顯著衰退,極端情況下會使得系統(tǒng)變得不穩(wěn)定,發(fā)生嘯叫。抑制聲反饋是擴聲系統(tǒng)亟待解決的問題。從改善房間聲學環(huán)境入手或在擴聲器中加入均衡器的傳統(tǒng)方法通常操作不便,并且費用較高。相位調(diào)制法和增益降低法是比較靈活的嘯叫抑制方法,但是在實時性、擴聲增益提高以及音質(zhì)損失之間很難獲得很好的平衡,且多為在嘯叫發(fā)生后進行檢測和處理,影響用戶的主觀聽覺感受。自適應嘯叫抑制法克服了相位調(diào)制法和增益降低法的缺點,能夠?qū)崿F(xiàn)實時處理,同時大幅提高系統(tǒng)增益,帶來較小的聲音失真,而且硬件成本較低。本論文以自適應嘯叫抑制法為主要研究對象,在深入分析自適應算法理論的基礎上,對自適應嘯叫抑制算法和去相關技術進行了討論和研究。論文首先介紹了自適應濾波器基本原理,并重點研究了LMS、NLMS、VMLMS以及VSNLMS算法。隨后研究了消除信號相關性的去相關技術,包括噪聲注入法、插入延時法、時變處理法和非線性處理法。接著闡述了利用自適應線性預測進行嘯叫抑制的原理。由于現(xiàn)階段對嘯叫抑制系統(tǒng)還沒有統(tǒng)一的測評標準,本文從系統(tǒng)性能、放大能力、音質(zhì)失真三方面采用多種評價標準結合的方法對嘯叫抑制進行評價。為了便于模擬嘯叫發(fā)生的聲場環(huán)境,論文搭建了MATLAB嘯叫抑制仿真平臺,并在此仿真平臺中對基于自適應線性預測的自適應嘯叫抑制算法及去相關技術進行仿真和分析實驗結果�？紤]到計算復雜度、擴聲增益和音質(zhì)失真等因素,論文選擇了NLMS算法和VMLMS算法進行DSP實現(xiàn)。硬件平臺選擇以TI公司的定點數(shù)DSP芯片(TMS320DM6437)為核心的EVM板,論文分別驗證了DSP算法在模擬聲場和真實聲場中的嘯叫抑制性能。為實現(xiàn)DSP與MATLAB仿真平臺的實時連接,本文設計了DSP與MATLAB的通信機制。經(jīng)過大量仿真測試和實際聲場測試,驗證了本文的嘯叫抑制方案能夠?qū)π盘枌崟r處理,嘯叫抑制效果較好,并且能獲得良好的聲音質(zhì)量。
[Abstract]:Acoustic feedback is a common problem in public sound reinforcement systems such as theatres, multimedia classrooms, meeting rooms and so on. It often makes the performance of audio reinforcement system decline significantly, and in extreme cases the system becomes unstable and howls. Acoustic feedback suppression is an urgent problem in sound reinforcement system. The traditional method of improving the acoustic environment of the room or adding equalizer to the loudspeaker is usually inconvenient and expensive. Phase modulation and gain reduction are flexible howling suppression methods, but it is difficult to obtain a good balance between real-time, acoustical gain improvement and sound quality loss, and most of them are detected and processed after howling occurs. Affect the user's subjective sense of hearing. The adaptive howling suppression method overcomes the shortcomings of the phase modulation method and the gain reduction method, and can achieve real-time processing. At the same time, it can greatly improve the system gain, bring less sound distortion and lower hardware cost. In this paper, the adaptive howling suppression method is taken as the main research object. On the basis of in-depth analysis of the adaptive algorithm theory, the adaptive howling suppression algorithm and de-correlation technology are discussed and studied. In this paper, the basic principle of adaptive filter is introduced, and the LMSN LMS VMLMS and VSNLMS algorithm are studied. Then the de-correlation techniques are studied, including noise injection method, insertion delay method, time-varying processing method and nonlinear processing method. Then the principle of roar suppression using adaptive linear prediction is described. Since there is no uniform evaluation standard for howling suppression system at present, this paper uses a variety of evaluation criteria to evaluate howling suppression from three aspects: system performance, amplification ability and sound quality distortion. In order to simulate the sound field environment, a MATLAB simulation platform for howling suppression is built in this paper. The simulation results of adaptive howling suppression algorithm based on adaptive linear prediction and de-correlation technology are simulated and analyzed in this simulation platform. Considering such factors as computational complexity, sound amplification gain and sound quality distortion, NLMS algorithm and VMLMS algorithm are selected for DSP implementation. The hardware platform is based on TI's fixed-point number DSP chip TMS320DM6437. the performance of DSP algorithm in simulated sound field and real sound field is verified in this paper. In order to realize the real-time connection between DSP and MATLAB, the communication mechanism between DSP and MATLAB is designed. Through a large number of simulation tests and actual sound field tests, it is verified that the proposed scheme can process the signal in real time, and the roar suppression effect is better and the sound quality can be obtained.
【學位授予單位】：電子科技大學
【學位級別】：碩士
【學位授予年份】：2014
【分類號】：TN912.3

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本文編號：2047853