Sound localization of concurrent and continuous speech sources in reverberant environment

J. Huang, N. Ohnishi, N. Sugie
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Abstract

This paper presents a model-based method for sound localization of concurrent and continuous speech sources in a reverberant environment. A new algorithm adopted from the echo-avoidance model of the precedence effect was used to detect the echo-free onsets by specifying a generalized pattern of impulse response. Fine structure time differences were calculated from the zero-crossing points in different microphones. They were integrated into an azimuth histogram by the restrictions between them. Two sound sources were localized in both an anechoic chamber and a normal room which has walls, floor and ceiling made of concrete. The time segment needed for localization was 0.5 to 2 seconds and the accuracy was a few degrees in both environments.
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混响环境中并发和连续声源的声音定位
提出了一种基于模型的混响环境下并发和连续语音源的声音定位方法。采用基于优先效应的回波回避模型的一种新算法,通过指定脉冲响应的广义模式来检测无回波触发。从不同传声器的过零点计算精细结构时间差异。通过它们之间的限制将它们整合到方位角直方图中。两个声源分别位于消声室和墙壁、地板和天花板由混凝土制成的正常房间。在两种环境下,定位所需的时间区间为0.5 ~ 2秒,精度为几度。
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Mixed nearfield/farfield beamforming: a new technique for speech acquisition in a reverberant environment Analysis of nonlinear and nonstationary processes in speech production Elimination of delay-free loops in discrete-time models of nonlinear acoustic systems Computational auditory scene analysis exploiting speech-recognition knowledge Towards a model of loudness recalibration
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