共查询到19条相似文献,搜索用时 134 毫秒
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为改善混响背景下传统匹配滤波算法效果不佳问题,在分析其非平稳性、有色性和非高斯性的基础上,提出了混合高斯时变自回归模型(Gaussian mixture Tvar Model,GTM),推导了模型公式及其参数求解方法,形成了GTM回波检测算法。为对混响特性及滤波效果进行定量描述进而验证算法性能,给出了一种定量衡量混响非平稳性、有色性、非高斯特性的滤波效果评价方法。通过实测混响分析表明,GTM模型能够较好地拟合实测混响的概率密度曲线和功率谱密度曲线,实现了混响背景下回波的有效检测并改善混响特性。 相似文献
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IR-1是仿真度极高的脉冲响应混响处理器插件,通过房间混响的声学特性系统采用时间线卷积的算法,可对混响时间、混响特性、混响效果参数、混响声阵包络进行调整,从而获得最具现场感的混响效果。 相似文献
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在相对封闭的声学环境中,由于受到混响的影响,麦克风阵列采集到的信号清晰度降低、甚至混淆不清。为了解决这一问题,文章在多通道线性预测(Multi-Channel Linear Prediction, MCLP)语音去混响的基础上,提出了一种改进的多通道线性预测(Multi-ChannelLinearPrediction,MCLP)方法即正交非负矩阵线性预测(Orthogonal Non-negative Matrix Factorization Multi-Channel Linear Prediction, ONMFMCLP)方法。该方法利用纯净语音的短时谱域的稀疏性,构建了基于正交的非负矩阵分解(Non-negative Matrix Factorization, NMF)的Kullback-Leibler(KL)问题,通过对矩阵求迹、利用梯度下降法给出迭代规则,进而改进了MCLP中目标信号矩阵的协方差估计。实验结果表明,相对于其他方法,ONMFMCLP方法具有更好的去混响效果。 相似文献
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隧道混响是隧道内列车噪声显著的关键因素,对混响特性进行预测分析有助于隧道内的降噪研究。采用脉冲响应积分法测试空场隧道混响时间,利用虚源法建立隧道壁面吸声系数数值计算模型,反推隧道壁面的吸声系数,将其输入给基于声线跟踪原理建立的隧道内声场响应预测分析模型,以混响时间和D/R比(Direct/Reverberant ratio)详细分析隧道内车体表面声场的混响特性。结果表明:隧道壁面吸声系数经验值与反推计算结果有较大差异,低于400 Hz频段经验值高于计算值,而高于400 Hz频段则反之。在车底响应面声源中心区域内直达声场强于混响声场,车底响应面的混响声场强度高于车顶响应面。混响时间均匀度从高到低的响应面分别为:车底、车顶、侧面,空场状态响应面上的平均混响时间明显高于有车状态下,其中车底响应面的最低。 相似文献
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隧道混响是隧道内列车噪声显著的关键因素,对混响特性进行预测分析有助于隧道内的降噪研究。采用脉冲响应积分法测试空场隧道混响时间,利用虚源法建立隧道壁面吸声系数数值计算模型,反推隧道壁面的吸声系数,将其输入给基于声线跟踪原理建立的隧道内声场响应预测分析模型,以混响时间和D/R比(Direct/Reverberant ratio)详细分析隧道内车体表面声场的混响特性。结果表明:隧道壁面吸声系数经验值与反推计算结果有较大差异,低于400 Hz频段经验值高于计算值,而高于400 Hz频段则反之。在车底响应面声源中心区域内直达声场强于混响声场,车底响应面的混响声场强度高于车顶响应面。混响时间均匀度从高到低的响应面分别为:车底、车顶、侧面,空场状态响应面上的平均混响时间明显高于有车状态下,其中车底响应面的最低。 相似文献
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0引言室内声学的研究始于混响现象的研究,而混响感是对混响现象的主观感知,它是厅堂音质中最基本的一种主观心理感觉。过去,常采用早期衰变时间EDT或混响时间RT来评价混响感,但随着研究的深入,人们发现这并不充分[1‐2]。由于混响时间的频率特性的差异,在不同频率的声能衰变情况下,会引起对混响的主观感知的差异。近年在听觉 相似文献
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CCD读出电路中低通滤波器在改善系统输出信噪比的同时,不可避免地会产生像素串扰,造成信息畸变.本文针对CCD读出电路中像素串扰造成的信息畸变问题进行理论分析,推导出信息畸变度与一阶低通模拟滤波器截止频率的关系.并针对视频模拟滤波电路中高阶滤波器的实现困难,提出一种数字补偿式一阶低通模拟滤波器的设计方法,该方法根据本文对信息畸变理论的分析,采用一阶低通滤波器,在满足信息畸变度的前提下,极大降低系统截止频率,达到了应用系统的带宽要求,从而可以在应用系统设计中以一阶滤波器替代高阶滤波器.并设计实验进行验证,证明了该方法的正确性与有效性. 相似文献
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Graph filtering, which is founded on the theory of graph signal processing, is
proved as a useful tool for image denoising. Most graph filtering methods focus on learning
an ideal lowpass filter to remove noise, where clean images are restored from noisy ones by
retaining the image components in low graph frequency bands. However, this lowpass filter
has limited ability to separate the low-frequency noise from clean images such that it makes
the denoising procedure less effective. To address this issue, we propose an adaptive
weighted graph filtering (AWGF) method to replace the design of traditional ideal lowpass
filter. In detail, we reassess the existing low-rank denoising method with adaptive
regularizer learning (ARLLR) from the view of graph filtering. A shrinkage approach
subsequently is presented on the graph frequency domain, where the components of noisy
image are adaptively decreased in each band by calculating their component significances.
As a result, it makes the proposed graph filtering more explainable and suitable for
denoising. Meanwhile, we demonstrate a graph filter under the constraint of subspace
representation is employed in the ARLLR method. Therefore, ARLLR can be treated as a
special form of graph filtering. It not only enriches the theory of graph filtering, but also
builds a bridge from the low-rank methods to the graph filtering methods. In the
experiments, we perform the AWGF method with a graph filter generated by the classical
graph Laplacian matrix. The results show our method can achieve a comparable denoising
performance with several state-of-the-art denoising methods. 相似文献
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声温法是基于声波与CT技术相结合的非接触测温法,声波信号飞渡时间的精确测量是温度场检测的关键环节。在复杂混响的背景环境下,传统的相关算法已无法克服卷积性干扰,可能会出现多相关峰值的处理结果。基于广义倒谱的相关分析算法在声温法温度检测应用中,利用倒谱运算将乘积同态系统与卷积同态系统变换成线性系统,从而分离乘积信号或卷积信号,并滤除乘积性干扰与卷积性干扰。理论与实验测量表明,与传统相关函数分析算法进行比较,新倒谱算法能够有效地克服混响的卷积干扰,并锐化峰值,从而准确估计出声信号的时延。 相似文献
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针对强混响背景下经典的最小均方误差(Least Mean Square,LMS)滤波算法难以有效地实现信混分离的问题,提出一种基于分数阶傅里叶变换的自适应LMS算法。首先将混响信号和自适应LMS滤波算法中的参考信号进行分数阶傅里叶变换,寻找最优变换域,并在分数阶域进行带通滤波,然后将得到的信号进行分数阶傅里叶反变换,最后将基于正态分布曲线的变步长LMS算法应用于此混响条件下进行滤波。仿真和海试数据验证结果表明,在信混比为0 dB的情况下,算法仍可以有效地滤除混响,使信混比提高6dB。 相似文献
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Zheng Y.R. Goubran R.A. El-Tanany M. 《IEEE transactions on instrumentation and measurement》2004,53(3):777-786
This paper proposes a near-field broadband adaptive beamforming scheme for intelligent computer telephony and teleconferencing applications, namely the nested microphone array with adaptive noise canceller (NMA-ANC). The NMA-ANC scheme incorporates an harmonically nested array with a nonuniformly subbanded multirate filter bank. Each subband array employs several near-field delay-filter-and-sum beamformers and an adaptive noise canceller (ANC). The proposed NMA-ANC is evaluated via a noise rejection experiment and dereverberation experiment performed in an anechoic chamber and a real conference room, respectively. The experiment data are recorded by a multichannel digital recording system developed using commercial off-the-shelf (COTS) equipments. A perceptual analysis/measurement system (PAMS) test is also carried out using a COTS digital speech level analyzer. The results of the experimental evaluation and PAMS test show that the proposed NMA-ANC scheme is able to improve the sound quality by adaptively rejecting multiple interfering signals and attenuating the reverberant noises and avoiding the desired signal cancellation. 相似文献
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传统的空时自适应处理(Space-Time Adaptive Processing,STAP)利用窄带混响和目标在空时平面的可分辨特征抑制窄带混响。对宽带混响和目标在空时平面分布特征进行研究,并通过理论推导,得出带宽对混响和目标空时分辨特征的影响公式。结果表明:宽带混响和目标在空时平面分布特征部分重叠,导致传统STAP效果不佳。在此基础上,借鉴STAP思想,并利用线性调频(Linear Frequency Modulation,LFM)信号在分数阶Fourier变换域上的聚焦性,分析了在空-分数阶Fourier域三维空间上抑制宽带LFM混响的可行性。 相似文献
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The paper presents a new concept for implementing tunable lowpass filters by employing slot resonators etched in the ground plane. When RF MEMS switches are used to short-circuit the slots in the ground plane, the effective length of the slots can be varied to achieve tunability at discrete frequencies. The concept is demonstrated by considering 4-slot lowpass filters. Continuous tuning is achieved by replacing switches with varactors as tuning elements. A varactor tuned lowpass filter was built and tested. Simple transmission line models for the proposed structure are also presented. The measured results are in good agreement with simulations confirming the validity of the proposed model. The experimental lowpass filters exhibit superior RF performance which consists of a low insertion loss and a large tuning range. The insertion loss is 0.6 dB for both the digital and the analogue tunable filters, while the achievable tuning range is 44% for the digital tunable filter and 22% for the analogue tunable filter. A much wider tuning range is obtained by combining digital and analogue tuning in one circuit 相似文献