基于嵌入式系統(tǒng)的實時聲音頻譜分析技術
發(fā)布時間:2018-12-10 13:43
【摘要】:聲壓頻譜分析是考慮人耳對不同頻率成分的聲音的感受的不同,進而通過傅里葉變換等獲得其準確頻譜特性的技術。聲壓頻譜分析是后續(xù)聲學分析的基礎,同時在聲學測量,噪聲污染,健康醫(yī)療,降噪減噪,故障診斷,國防建設等中都具有重要的應用。本課題從理論分析到硬件及算法設計介紹了基于嵌入式系統(tǒng)的手持式聲壓分析平臺的實現(xiàn)過程。提出了一套精度較高,運算量較小,實時性較好,可操作性較強的手持式聲學頻譜分析方案。 本課題對基于嵌入式系統(tǒng)的聲壓頻率計權、頻譜分析進行了深入的分析與研究,首先介紹了基于卷積的頻率計權實現(xiàn)及基于傅里葉變換、快速傅里葉變換的聲壓頻譜分析方法?紤]卷積運算的運算量比較大,本文提出了基于重疊相加DFT/IDFT的頻率計權優(yōu)化算法,并分析了兩者的算法復雜度及最優(yōu)幀長選擇標準。針對Matlab庫函數(shù)設計的頻率計權濾波器與本課題目標計權濾波器頻響相差較大,介紹了一種基于DFT/IDFT的頻率計權濾波器的設計方法。考慮FFT變換是基于復數(shù)域的,研究了聲壓信號的FFT/IFFT加速實現(xiàn)算法。最后對本課題的實驗平臺的實現(xiàn)進行了系統(tǒng)分析,介紹了主要的硬件構成和相應的驅動程序開發(fā),分析了輸入輸出通道的校準方法以及在快速傅里葉變換過程中需要使用到旋轉因子,位反系數(shù)和分離因子的計算方法。 對本課題的研究方案,分別在Matlab上進行仿真分析和嵌入式平臺上進行實際測試,結果表明,基于重疊相加DFT變換的頻率計權實現(xiàn)方法與基于卷積的實現(xiàn)方法結果一致,而且隨著濾波器系數(shù)的增大前者性能快速提升,對計權后的信號進行時域積分,證明本課題的研究方法的計算結果滿足GB3240-1982一級標準。對正弦信號進行頻譜分析,也可以得到近似脈沖信號的主瓣,較小的邊瓣和過渡。
[Abstract]:The sound pressure spectrum analysis is a technique that takes into account the different perception of the human ear to the sound with different frequency components, and then obtains the exact spectrum characteristics by Fourier transform and so on. Sound pressure spectrum analysis is the basis of subsequent acoustic analysis, and it has important applications in acoustic measurement, noise pollution, health care, noise reduction, fault diagnosis, national defense construction and so on. From theoretical analysis to hardware and algorithm design, this paper introduces the realization process of handheld sound pressure analysis platform based on embedded system. In this paper, a handheld acoustic spectrum analysis scheme with high precision, low computation, good real-time and high maneuverability is proposed. In this paper, the sound pressure frequency measurement and frequency spectrum analysis based on embedded system are deeply analyzed and studied. Firstly, the realization of frequency weighting based on convolution and the method of sound pressure spectrum analysis based on Fourier transform and fast Fourier transform are introduced. Considering the complexity of convolution operation, a frequency weight optimization algorithm based on overlapping additive DFT/IDFT is proposed, and the complexity of the two algorithms and the optimal frame length selection criteria are analyzed. Aiming at the difference between the frequency response of the frequency weighting filter designed by Matlab library function and that of the target weighted filter, a design method of the frequency weighted filter based on DFT/IDFT is introduced. Considering that FFT transform is based on complex domain, the FFT/IFFT acceleration algorithm of sound pressure signal is studied. Finally, the realization of the experiment platform is analyzed systematically, and the main hardware structure and the corresponding driver development are introduced. The calibration method of input and output channels and the calculation method of rotation factor, bit inversion coefficient and separation factor are analyzed. The results show that the frequency weighting method based on overlapping plus DFT transform is consistent with that based on convolution. Moreover, with the increase of filter coefficient, the performance of the former is improved rapidly, and the time domain integral of the weighted signal is carried out, which proves that the calculation results of this research method meet the GB3240-1982 first order standard. The main lobe, small sidelobe and transition of the approximate pulse signal can also be obtained by spectrum analysis of the sinusoidal signal.
【學位授予單位】:浙江大學
【學位級別】:碩士
【學位授予年份】:2014
【分類號】:TB526
[Abstract]:The sound pressure spectrum analysis is a technique that takes into account the different perception of the human ear to the sound with different frequency components, and then obtains the exact spectrum characteristics by Fourier transform and so on. Sound pressure spectrum analysis is the basis of subsequent acoustic analysis, and it has important applications in acoustic measurement, noise pollution, health care, noise reduction, fault diagnosis, national defense construction and so on. From theoretical analysis to hardware and algorithm design, this paper introduces the realization process of handheld sound pressure analysis platform based on embedded system. In this paper, a handheld acoustic spectrum analysis scheme with high precision, low computation, good real-time and high maneuverability is proposed. In this paper, the sound pressure frequency measurement and frequency spectrum analysis based on embedded system are deeply analyzed and studied. Firstly, the realization of frequency weighting based on convolution and the method of sound pressure spectrum analysis based on Fourier transform and fast Fourier transform are introduced. Considering the complexity of convolution operation, a frequency weight optimization algorithm based on overlapping additive DFT/IDFT is proposed, and the complexity of the two algorithms and the optimal frame length selection criteria are analyzed. Aiming at the difference between the frequency response of the frequency weighting filter designed by Matlab library function and that of the target weighted filter, a design method of the frequency weighted filter based on DFT/IDFT is introduced. Considering that FFT transform is based on complex domain, the FFT/IFFT acceleration algorithm of sound pressure signal is studied. Finally, the realization of the experiment platform is analyzed systematically, and the main hardware structure and the corresponding driver development are introduced. The calibration method of input and output channels and the calculation method of rotation factor, bit inversion coefficient and separation factor are analyzed. The results show that the frequency weighting method based on overlapping plus DFT transform is consistent with that based on convolution. Moreover, with the increase of filter coefficient, the performance of the former is improved rapidly, and the time domain integral of the weighted signal is carried out, which proves that the calculation results of this research method meet the GB3240-1982 first order standard. The main lobe, small sidelobe and transition of the approximate pulse signal can also be obtained by spectrum analysis of the sinusoidal signal.
【學位授予單位】:浙江大學
【學位級別】:碩士
【學位授予年份】:2014
【分類號】:TB526
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