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M. Ashrafand
R. Mahmud


A fast processing approach is presented to convert the digital data gathered at lower sampling-rate into high sampling rate by utilizing the sampling rate changes algorithms using interpolation. A high speed low complexity interpolation filter is implemented at a very low hardware cost using the polyphase form, distributed arithmetic and lookup-tables. The proposed algorithm is used to processes the input data in parallel using a few shift and add operations, which make it very fast. The method allows for the multiplication of the original information stored during digitization of the analog signal at lower computational cost. Therefore, it is an interesting alternative for more sophisticated methods of performance enhancement of sampled analogue signals.

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