本文選題：麥克風(fēng)陣列 + 聲源定位�。� 參考：《南京大學(xué)》2015年碩士論文

【摘要】：在陣列信號處理領(lǐng)域,基于麥克風(fēng)陣列的聲源定位技術(shù)是一項研究熱點,其在軍事和民用等領(lǐng)域都得到了廣泛應(yīng)用。在實際應(yīng)用中,由于存在環(huán)境噪聲和混響干擾,尤其是在室內(nèi)環(huán)境中(如小型會場、教室、辦公室),實現(xiàn)精確定位是當(dāng)前聲源定位技術(shù)的主要難點。此外,聲源定位還存在運算量較大,系統(tǒng)成本較高等問題。針對這些問題,本文致力于室內(nèi)環(huán)境下麥克風(fēng)陣列聲源定位系統(tǒng)的算法研究和系統(tǒng)實現(xiàn)工作,重點研究了基于時延估計的聲源定位算法,提出了基于GCC改進算法的分數(shù)時延估計算法,并在實際環(huán)境中搭建了定位系統(tǒng),采集了實測數(shù)據(jù),驗證了算法的有效性。本文主要工作如下：1.對當(dāng)前三種聲源定位算法進行理論分析,基于時延估計的聲源定位算法具有算法實用性高,實時性好等優(yōu)點,選擇其作為本文的研究重點；2.分析了麥克風(fēng)陣列的空間模型(遠場模型和近場模型)以及信號接收模型(理想模型和實際模型),根據(jù)IMAGE模型,給出了房間沖擊響應(yīng)的表達式和模型仿真；3.對時延估計算法進行系統(tǒng)研究,重點介紹了廣義互相關(guān)和自適應(yīng)最小均方兩種時延估計算法；在廣義互相關(guān)算法的基礎(chǔ)上,針對低信噪比、高混響的語音環(huán)境,利用混響能量比例系數(shù),提出了一種基于GCC改進算法的分數(shù)時延估計算法,進一步提高了時延估計的精確度；4.對幾何定位算法做了系統(tǒng)的研究,重點研究了最大似然估計算法和最小二乘估計算法；5.設(shè)計開發(fā)一套基于十字形麥克風(fēng)陣列的聲源定位系統(tǒng),詳細介紹了聲源定位系統(tǒng)整體結(jié)構(gòu),并對整個定位系統(tǒng)做了相關(guān)測試實驗,分析了可能導(dǎo)致誤差產(chǎn)生的原因。
[Abstract]:In the field of array signal processing, sound source location based on microphone array is a research hotspot, which has been widely used in military and civilian fields. In practical applications, due to the presence of environmental noise and reverberation interference, especially in the indoor environment (such as small meeting room, classroom, office), accurate location is the main difficulty of the current sound source location technology. In addition, there are many problems in sound source location, such as large computation and high system cost. In order to solve these problems, this paper focuses on the algorithm research and system implementation of microphone array sound source location system in indoor environment, and focuses on the acoustic source location algorithm based on time delay estimation. A fractional time delay estimation algorithm based on GCC improved algorithm is proposed, and a positioning system is built in the actual environment, and the measured data are collected to verify the validity of the algorithm. The main work of this paper is as follows: 1. Based on the theoretical analysis of three current acoustic source location algorithms, the acoustic source location algorithm based on time delay estimation has the advantages of high practicability and good real-time performance, so it is chosen as the focus of this paper. The space model (far-field model and near-field model) and the signal receiving model (ideal model and actual model) of microphone array are analyzed. According to the image model, the expression of room impulse response and the simulation of the model are given. Based on the generalized cross-correlation algorithm and the adaptive minimum mean square algorithm, the generalized cross-correlation algorithm and the adaptive least mean square algorithm are introduced systematically, and then, for the low signal-to-noise ratio and high reverberation environment, the time delay estimation algorithm is studied systematically. A fractional time delay estimation algorithm based on GCC improved algorithm is proposed based on reverberation energy proportional coefficient, which improves the accuracy of time delay estimation. The geometric location algorithm is studied systematically, and the maximum likelihood estimation algorithm and the least square estimation algorithm are studied emphatically. A sound source location system based on cross-shaped microphone array is designed and developed. The overall structure of sound source positioning system is introduced in detail.
【學(xué)位授予單位】：南京大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2015
【分類號】：TN641;TN911.7

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本文編號：2101295