| Summary: | 碩士 === 國立成功大學 === 電機工程學系碩博士班 === 96 === The aim of sound localization system (SLS) is to identify the direction of sound source. SLS is often necessary for the applications of toy and robot hearing, etc. The principle theory of SLS is estimating time difference (TD) of the signals received by microphone pair. And then, the time difference information converts into the directional angle. An SLS consists of the input sound detection (ISD) processing and the source localization estimation (SLE) processing. The ISD detects whether the input signal is the sound desired signal or noise. On the other hand, the SLE finds the direction and angle of source.
To date, a great deal of effort has been devoted to providing better sound localization systems. Most of them focus on near-field identification. However, when the distance between sound source and receiver becomes longer, the accuracy decreases more. In many literatures, the distance ranges of identification for SLS are between 1 to 2 m; these ranges may not fit application requirements. For this reason, the purpose of this study is to extend the distance ranges to 5 m. The development of SLS should be low cost, low design complexity and small hardware area. We focus on time domain signal processing and construct a far-field sound localization system base on the average magnitude difference function (AMDF). In the reverberation environment, echo will occur on far-field identification. We only extract the onset signals in order to alleviate the reverberant problem. The far-field SLS is verified on the PC and further is implemented on the field programmable gate array (FPGA). Compared with other literatures, our far-field SLS is implemented on a single FPGA chip. The hardware area only occupies about 200,000 logic gates and the distance of identification owns 5 m. In our experiments, the performance achieves almost 90% accuracy for clap within ±5° error.
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