人机交互中的声源定位与增强方法

基于人机交互的实际应用场景，提出了一种交互目标声源三维定位与语音增强算法。该算法首先在广义相关法的基础上提出一种声达时延差的估计方法，通过由6个麦克风构成的平行均匀线阵接收模型，实现对目标声源的三维定位；然后在交互目标声源定位的基础上，通过调整时延波束叠加的权值，实现对目标语音的增强。仿真实验结果表明，文中提出的算法能够准确定位目标声源并对目标语音进行有效增强。在信噪比大于1.5 dB 的环境中，该算法可使目标声源达到98%以上定位精度，5 dB 以上信噪比改善，同时算法运算量小且易于硬件实现。

Acoustic Source Localization and Speech Enhancement in Human-Computer Interaction

Focusing on the human computer interaction(HCI) real scenarios, a method for acoustic source three-dimensional localization and speech enhancement is proposed. Combining with the receiver model of six microphone parallel uniform linear array (ULA), the target acoustic source is located in three dimensions based on the method of time-difference of arrival (TDOA) estimation improved from generalized cross correlation (GCC).On the basis of positioning the target acoustic source, the target speech enhancement is implemented by the method of delay-and-sum beamforming (DSBF). Simulation results show that the method can position the target acoustic source accurately and enhance the target speech effectively. In the context of SNR greater than 1.5 dB,the positioning accuracy of the target acoustic source can reach more than 98% and improvement of SNR can reach 5 dB with less computation cost and easier hardware implementation.