| Summary: | 碩士 === 龍華科技大學 === 電子系碩士班 === 94 === The adaptive filter is an important tool in digital signal processing. Depending on the changing of system conditions, an adaptive filter can be designed to estimate the parameters of the system and to adjust accordingly. Because of this advantage and its simplicity, the time-domain adaptive LMS algorithm has attracted a lot of attention and become one of the most widely used techniques in the past decade. However, because its converging speed is slow and there is no generally applicable rule in deciding the step size in each iteration, this technique is not very easy to implement. For this reason, many researchers have devoted their efforts in solving these problems.
This paper derives an algorithm to estimate the step size of the adaptive filter and simulate the behavior of the adaptive noise canceller in real time. The proposed ANC has two adaptive filters: a main filter (MF) and a subfilter (SF). The signal-to-noise ratio (SNR) of input signals is estimated using the SF. To reduce signal distortion in the output signal of the ANC, a step size for coefficient update in the MF is controlled according to the stimated SNR. Moreover, comparative studies on numerical experiments demonstrate the effectiveness of the proposed algorithm.
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