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麦克风阵列的自适应波束形成Matlab仿真,麦克间距和信号带宽可调,采用最小均方自适应波束形成算法-microphone array of adaptive beamforming Matlab simulation, and signal spacing Mike adjustable bandwidth, using LMS adaptive beamforming algorithms
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麦克风阵列的TLS自适应波束形成算法仿真,麦克间距和输入信号带宽可调,通过调整参数达到需要的输出-microphone array of adaptive beamforming TLS algorithm simulation, Mike spacing and adjustable input signal bandwidth. by adjusting the parameters of the needs of the output
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广义旁瓣抵消器(GSC)是麦克风阵列语音增强中被广泛采用的一种自适应波束形成算法-generalized sidelobe canceller (GSC) is the microphone array speech enhancement which has been widely used in an adaptive beamforming Algorithm
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一种新的基于麦克风阵列的近场声源定位和语音分离算法,它结合双波束二维定位和近场最小方差波束形成技术在阵列近场范围内实现声源定位和语音分离。-based on a microphone array of near-field acoustic source localization and speech separation algorithm, It combines two-beam two-dimensional positioning and near-field minimum vari
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基于麦克风阵波束形成的谱相减方法在语音增强中的应用-Microphone array beamforming based on the spectral subtraction method speech enhancement application
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麦克风阵列的TLS自适应波束形成算法仿真,麦克间距和输入信号带宽可调,通过调整参数达到需要的输出-TLS microphone array adaptive beamforming algorithm simulation, Mike spacing and input signal bandwidth is adjustable by adjusting the parameters to achieve the required output
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争对移动声源采用波束形成进行语音增强,提出一种约束子带波束形成算法。其波束形成器基于一个软约束,其目的是要使波束指向特定的区域即声源方向。而其核心在于首先要进行声源定位,获得尽量准确的方位信息,然后构造软约束条件,用于波束形成。且在此过程中不断跟踪声源的移动情况。在构造的约束条件中,需要知道声源的二维信息,即与麦克风阵列的距离和方向角,“软”体现在对距离和方向角的确定都是在一定范围内的,有待进一步更正。-Between the mobile sound source using beamform
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近场麦克风阵列波束形成方法研究.kdh
硕士毕业论文-Near-field microphone array beamforming method. Kdh Master' s Thesis
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麦克风阵语音增强中波束形成后维纳滤波的MATLAB代码-Microphone Array Speech Enhancement Beamforming MATLAB code after the Wiener filtering
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麦克风阵语音增强中时延估计的MATLAB代码-Microphone Array Speech Enhancement in the time-delay estimation of the MATLAB code
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麦克风阵列的一些波束形成算法打包 可以很好的理解波束形成原理-Some of the microphone array beamforming algorithms can be packaged to form a good understanding of beam theory
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基于均匀圆形麦克风阵列模型的延时-求和波束赋形成-Delay-Sum Beamforming based on the uniform circular microphone array model
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基于GSC结构的多麦克风数字助听器的自适应波束形成算法_-Multiple microphone digital hearing aid based on GSC structure of adaptive beamforming algorithm
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麦克风阵 + 波束形成 后Wiene滤波-Microphone array+ beamforming after Wiene filtering
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程序描述的是等边三角形麦克风阵列语音增强的程序,采用的是延时求和波束形成的方法,同时实现了声源定位功能。-Procedure is described in an equilateral triangle microphone array speech enhancement procedure, using the delay sum beamforming method, while achieving a sound source localization capabilities.
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基于麦克风阵列的自适应波束形成算法.它是基于SMI自适应算法-Microphone array based adaptive beamforming algorithm which is based on the SMI adaptive algorithm
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方向性_基于GSC结构的多麦克风数字助听器的自适应波束形成算法_高杰胡广书-An adaptive Beamforming Algorithm in GSC Structure for Multi2
Microphone Digital Hearing Aids
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麦克风阵列mvdr算法,有音源和输出,能直接运行(Microphone array MVDR algorithm, with sound source and output, can run directly)
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8阵元均匀线阵的波束形成,波束宽度与波达方向及阵元数的关系(ULA-beamforming,8 microphone)
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波速成型算法、麦克风阵列语音信号技术相关文档以及webrtc相关文档(Wave speed forming algorithm, microphone array voice signal technology related documents and webrtc related documents)
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