Digital signals are often available in a wide variety of formats and with different sampling rates. : For maximum flexibility in processing such signals it is important to be able to digitally change the sampling rate of an incoming signal to almost any desired rate. The theory is reviewed, and some practical implementations of digital systems are discussed which can change the sampling rate of a signal. The Nyquist sampling theorem is the basis for all sampling rate conversion techniques. It is shown how a simple, straightforward application of the sampling theorem leads to the canonical system for sampling rate conversion. Considerations in the implementation of a sampling rate converter, including structures, filter designs, ad cascading techniques, are briefly discussed. Finally an example is given of a signal processing system based on sampling rate conversion.
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