基于改進(jìn)GCC-PHAT算法的麥克風(fēng)陣列聲源定位技術(shù)
本文選題:時(shí)延估計(jì) + 廣義互相關(guān); 參考:《遼寧工程技術(shù)大學(xué)》2017年碩士論文
【摘要】:隨著人民生活水平的日益提高和科學(xué)技術(shù)的的進(jìn)步,服務(wù)型機(jī)器人的應(yīng)用也在逐漸的推廣。針對(duì)比較貼近人類日常生活的服務(wù)型機(jī)器人來說,更好的實(shí)現(xiàn)人機(jī)交互就顯得十分必要。語(yǔ)音識(shí)別已經(jīng)在機(jī)器人上得到了應(yīng)用,聲源定位功能的開發(fā)也將是一個(gè)具有重要意義的研究方向。正是在這樣一個(gè)背景下,本文對(duì)服務(wù)型機(jī)器人上的聲源定位方法進(jìn)行了研究。提出了相應(yīng)的解決方法,有望提高其智能化水平。本文的主要研究?jī)?nèi)容有以下幾個(gè)方面。(1)分析比較了常用的幾種聲源定位方法,最后選擇了易于實(shí)現(xiàn)的基于聲信號(hào)到達(dá)時(shí)間差的聲源定位方法作為本文的研究主線。(2)對(duì)信號(hào)預(yù)處理階段開展了細(xì)致的研究;拘盘(hào)模型進(jìn)行了分類。針對(duì)語(yǔ)音段與非語(yǔ)音段的劃分,采用了行之有效的語(yǔ)音端點(diǎn)檢測(cè)方法-頻帶方差法,可以有效去除非語(yǔ)音段,減小了計(jì)算復(fù)雜度。由于一般的帶通濾波只能濾除設(shè)定頻帶范圍外的噪聲,對(duì)于頻帶內(nèi)疊加的噪聲無(wú)可奈何,所以提出了基于改進(jìn)譜減法的語(yǔ)音增強(qiáng)方法。(3)系統(tǒng)分析了幾種時(shí)延估計(jì)方法,對(duì)它們的工作原理進(jìn)行了比較細(xì)致的研究,并進(jìn)行了仿真對(duì)比。最后選定GCC-PHAT作為本文的時(shí)延估計(jì)方法,并改進(jìn)了它的時(shí)延估計(jì)性能,使它在信噪比較低的情況下時(shí)延估計(jì)的準(zhǔn)確性以及可靠性得到提高,為下一步的位置估計(jì)提供可靠的時(shí)延值。(4)介紹了兩大類定位估計(jì)方法:幾何定位法和目標(biāo)函數(shù)搜索法。在傳統(tǒng)幾何定位法的基礎(chǔ)上提出了一種基于七元麥克風(fēng)立體十字陣列的定位方法,快速的縮小了定位的范圍,有效排除了模糊解,提高了定位的成功率。(5)對(duì)基于聲信號(hào)到達(dá)時(shí)間差的聲源定位方法進(jìn)行理論分析以后,又進(jìn)行了實(shí)驗(yàn)驗(yàn)證。通過麥克風(fēng)陣列采集語(yǔ)音信號(hào),經(jīng)處理器對(duì)數(shù)據(jù)打包處理后通過串口送入電腦端,運(yùn)用MATLAB軟件對(duì)采集到的數(shù)據(jù)進(jìn)行分析處理,最終得出聲源的估計(jì)位置。結(jié)果表明本文所采用的方法能夠定位出聲源的位置,且精度較高,能夠滿足一定的實(shí)際需要。
[Abstract]:With the improvement of the people's living standard and the progress of science and technology, the application of the service robot is also gradually popularized. For the service robot which is close to human daily life, it is necessary to realize the human-computer interaction better. The speech recognition has been applied to the robot and the sound source positioning function is used. Development will also be an important research direction. Under such a background, this paper studies the sound source localization method on the service robot. It puts forward the corresponding solutions and is expected to improve its level of intelligence. The main contents of this paper are as follows. (1) analysis and comparison of several common methods are used. The sound source localization method is selected as the main line of this paper. (2) a detailed study of the signal preprocessing stage is carried out. The basic signal model is classified. The effective voice endpoint detection side is adopted for the division of the speech and non speech segments. The method of frequency band variance can effectively remove the non speech segment and reduce the computational complexity. Because the general bandpass filter can only filter the noise outside the range of the frequency band, the noise overlay in the frequency band is helpless, so a speech enhancement method based on the improved spectral subtraction is proposed. (3) several time delay estimation methods are analyzed systematically. The working principle is studied carefully and the simulation comparison is carried out. Finally, GCC-PHAT is selected as the time delay estimation method in this paper, and its delay estimation performance is improved, which makes it improve the accuracy and reliability of time delay estimation under low signal noise comparison, and provides reliable delay for the next position estimation. (4) (4) two kinds of location estimation methods are introduced: geometric positioning method and target function search method. Based on the traditional geometric positioning method, a positioning method based on seven yuan microphone stereoscopic array is proposed, which reduces the range of positioning quickly, effectively eliminates the fuzzy solution, and improves the success rate of positioning. (5) sound signals based on sound signals. After the theoretical analysis of the sound source location method of the arrival time difference, the experimental verification is carried out. The speech signal is collected through the microphone array. After the processing of the data, the data is sent to the computer side through the serial port through the processor. The data collected by the MATLAB software is analyzed and processed, and the estimation position of the sound source is finally obtained. The method adopted in this paper can locate the location of the sound source, and the accuracy is high, which can meet certain practical needs.
【學(xué)位授予單位】:遼寧工程技術(shù)大學(xué)
【學(xué)位級(jí)別】:碩士
【學(xué)位授予年份】:2017
【分類號(hào)】:TP242;TN912.3
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