In digital audio systems, the digital-to-analog (D/A) converter as well as the lowpass filter play an important role. Current trend of implementing the filter uses digital oversampling (or interpolating) filter rather a pure analog filter due to the better performance and the ease of integration in VLSI. However, the direct implementation of the digital oversampling filters usually requires very high order FIR filters to achieve the desired specification. Such a high order filter requires high speed circuitry to perform the multiplications. In this paper, we show that by utilizing the property of half-band filters and a two-stage implementation, the order of the oversampling filter which operating at an oversampling factor of 4 can be reduced and hence the required computation and memory can also be cut down significantly.
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