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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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语音分离的研究在语音通信、声学目标检测、声音信号增强等方面有着重要的理论意义和实用价值。而将语音分离技术应用到智能机器人中。让机器人具有智能的听觉，实现声源定位和分离，确定说话人个数，进行人机对话等方面更具有广阔的应用前景。 -Separation of speech in voice communications, acoustic target detection, sound signal enhancement, and so has important theoretical sign
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声源定位，根据近场二维宽带MUSIC算法，自己编写的程序！-The speech source localization base on microphone array. Matlab program on near-field 2D music algorithm.
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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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声源定位程序，语音信号与房间脉冲响应进行卷积得到混响语音信号，再通过互相关时延估计得到定位-Sound source localization procedure, the speech signal and the room impulse response are convoluted to get the reverberation speech signal, and then through the correlation time delay estimation to obtain
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