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1.
应用由111个传声器组成的平面传声器阵列对当前流行的民用客机进场着陆过程中的机体噪声源进行了实验测量,本对七架窄体客机和七架宽体客机的起落架噪声进行了分析,得到了起落架噪声的频谱特性、指向特性和声级变化。研究发现,起落架噪声的频谱是由宽频随机噪声与一些较为明显的单噪声源组成,起落架噪声的指向性类似于一个水平放置的偶极子。不同飞机起落架噪声的声级相差较大,这说明可以通过重新结构设计降低起落架噪声。  相似文献
2.
基于波束形成的多类型多声源定位研究*   总被引:1,自引:1,他引:0       下载免费PDF全文
肖栋  向阳  卓瑞岩  王磊 《应用声学》2017,36(3):220-227
为实现空压机多噪声源的准确定位,仿真对比了多种近场球面波多声源定位算法。基于时域波束形成,研究了相同声源平面、不同声源频率、不同声源纵向距离、不同声源强度下多声源定位以及声源频率、声源纵向距离和声源强度多因素联合的多声源定位仿真方法,模拟了更接近实际的噪声源类型。基于频域波束形成,仿真研究了1400 Hz,2400 Hz,3400 Hz,4400 Hz的多声源。分别利用互功率谱波束形成和除自谱的互功率谱波束形成,仿真研究了相干声源和不相干声源。开发了阵列声成像测试平台,运用频域波束形成和功率谱波束形成对空压机进行了定位试验研究。结果表明,1400 Hz下空压机的主要噪声源是气缸盖、空气滤清器和曲轴附近的机体,这可为空压机减振降噪改进设计提供依据。  相似文献
3.
In microphone arrays application, it is a difficult task to accurately and fast localize sound source in a noisy, reverberant environment. In order to solve this problem, many approaches have been presented. Among them, the steered response power-phase transform weighted (SRP–PHAT) source localization algorithm has been proved robust. However, SRP–PHAT requires high computation cost for searching a large location space. To overcome this shortcoming, an improved SRP–PHAT will be presented that reduces a two-dimension searching space into a couple of one-dimension ones by using an orthogonal linear array. In this method, the parameters of direction of arrival (DOA) are separated. The main computation can be carried out independently in two one-dimension spaces, thus the computational load will be greatly cut down. Simulations show that there is no loss in accuracy in the proposed method.  相似文献
4.
The recent expansion of French tram networks and the related local residential complaints mean that a better knowledge of the situations leading to negative reactions from the local inhabitants is required. Hence a research project has been conducted to evaluate and describe noise and vibration emission of trams as well as the perception by the local residents. This paper investigates tram noise emission on common straight track sections, involving two vehicle scales. First the acoustic power and the mean vertical directivity of the total tramset is assessed using an arc of microphones. Then the localisation and the analysis of the main noise sources are performed by means of a cross array during the tram pass-by. Two tram types representing two generations of French rolling stock, both running on two sites with distinct track characteristics, have been investigated considering the effect of speed, tram type, and track type on the noise source contributions and spectral features. Most sources are located in the lower part of the trams, mainly related to rolling noise, with a strong dependence on speed and track type. The tram type dependency, although globally of second importance, influences greatly the noise spectral distribution and behaviour. The HVAC was the only roof-mounted source which could be detected; its contribution towards building storeys becomes significant in configurations of low rolling noise. A tram noise emission model based on the various noise sources has been developed.  相似文献
5.
Xinwang Wan 《Applied Acoustics》2010,71(12):1126-1131
Sound source localization is essential in many microphone arrays application, ranging from teleconferencing systems to artificial perception in a reverberant noisy environment. The steered response power (SRP) using the phase transform (SRP-PHAT) source localization algorithm has been proved robust, however, the performance of the SRP-PHAT algorithm degrades in highly reverberant noisy environment. Though the SRP-based maximum likelihood localizers are more robust than SRP-PHAT, they have the drawback of requiring noise variance to be estimated in a silent room. This paper presents an improved SRP-PHAT algorithm based on principal eigenvector. Sound source location is estimated from the principal eigenvector computed from the frequency-domain correlation matrix. Using both simulated and real data, we show that the proposed algorithm achieves higher source localization accuracy compared to the SRP-PHAT algorithm.  相似文献
6.
In this paper, a hybrid post-filter for microphone arrays with the assumption of a diffuse noise field is proposed to suppress correlated as well as uncorrelated noise. In the proposed post-filter, a modified Zelinski post-filter, which is estimated using the signals on the microphone pairs on which noises are uncorrelated by considering the correlation characteristics of noise impinging on different microphone pairs, is applied to the high frequencies to suppress spatially uncorrelated noise; a single-channel Wiener post-filter is applied to the low frequencies for cancellation of spatially correlated noise. In theory, the proposed post-filter is a Wiener post-filter. In practice, experiments using multi-channel recordings were conducted, and experimental results demonstrate the usefulness and superiority of the proposed post-filter compared to other post-filters using speech quality measures and speech recognition rate.  相似文献
7.
In this paper a novel method for tracking an active speaker in a noisy and reverberant environment by means of a spatially distributed microphone array is presented. Firstly, a sound source localization algorithm based on time delays of arrival (TDOA) in microphone pairs provides observed position estimates. Then these remarkably noisy estimates are filtered by a multiple model Kalman filter (MMKF) in order to obtain a smoothed trajectory of the speaker’s movement. Compared with the traditional Kalman filter (KF), simulated results prove the MMKF is more robust and effective in noisy environments.  相似文献
8.
9.
The traditional microphone configuration used to measure room impulse responses (IRs) according to ISO 3382:2009 is an omnidirectional and figure-8 microphone pair. IRs measurements were taken in a 2500-seat auditorium to determine how the results from a spherical microphone array (an mh acoustics Eigenmike-em32) compare to those from the traditional microphone setup (a Brüel & Kjær Type-4192 omnidirectional microphone and a Sennheiser MKH30 figure-8 microphone). Measurements were obtained at six receiver locations, with three repetitions each in order to first evaluate repeatability. The metrics considered in this study were: reverberation time (T30), early decay time (EDT), clarity index (C80), strength (G), lateral energy fraction (JLF) and late lateral energy level (LJ). Before calculating these quantities, the IRs were filtered to equalize the frequency response of the microphones and sound source. For the spherical array measurements, the omnidirectional (monopole) and figure-8 (dipole) patterns were extracted using beamforming. In terms of repeatability, the average standard deviation of the three measurements at each receiver location averaged across all metrics, receivers, and octave bands was found to be 0.01 just noticeable differences (JNDs). The analysis comparing the measurements from the two microphone configurations yielded differences which were less than 1 JND for the majority of metrics, with a few exceptions of EDT and C80 slightly above 1 JND. Based on this case study, these results indicate that spherical microphone arrays can be used to obtain valid room IR measurements, which will allow for the development of new metrics utilizing the higher spatial resolution made possible with spherical arrays.  相似文献
10.
Due to their small size, differential microphone arrays (DMAs) are very attractive. Moreover, they have been effective in combating noise and reverberation. Recently, a new class of DMAs of different orders have been developed with the MacLaurin’s series and the frequency-independent patterns. However, the MacLaurin’s series does not approximate well the exponential function, which appears in the general definition of the beampattern, when the intersensor spacing is not small enough. To circumvent this problem, we propose in this paper to approximate the exponential function with the Jacobi–Anger expansion. Based on this approximation and the frequency-independent Chebyshev patterns, we derive first-, second-, and third-order DMAs. Furthermore, in order to improve the robustness of DMAs against white noise amplification, we propose to use more microphones combined with minimum-norm filters. It is also shown that the Jacobi–Anger expansion is optimal from a mean-squared error perspective. Simulations are carried out to evaluate the performance of the proposed DMAs.  相似文献
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