### Downsampling and Aliasing

The*downsampling*operator selects every sample of a signal:

(3.32) |

The

*aliasing theorem*states that downsampling in time corresponds to

*aliasing*in the frequency domain:

(3.33) |

where the operator is defined as

(3.34) |

for . The summation terms for are called

*aliasing components*. In

*z*transform notation, the operator can be expressed as [287]

(3.35) |

where is a common notation for the primitive th root of unity. On the unit circle of the plane, this becomes

(3.36) |

The frequency scaling corresponds to having a sampling interval of after downsampling, which corresponds to the interval prior to downsampling. The aliasing theorem makes it clear that, in order to downsample by factor without aliasing, we must first lowpass-filter the spectrum to . This filtering (when ideal) zeroes out the spectral regions which alias upon downsampling. Note that any rational sampling-rate conversion factor may be implemented as an upsampling by the factor followed by downsampling by the factor [50,287]. Conceptually, a stretch-by- is followed by a lowpass filter cutting off at , followed by downsample-by- ,

*i.e.*,

(3.37) |

In practice, there are more efficient algorithms for sampling-rate conversion [270,135,78] based on a more direct approach to

*bandlimited interpolation*.

#### Proof of Aliasing Theorem

To show:(3.38) |

where we have chosen to keep frequency samples in terms of the original frequency axis prior to downsampling,

*i.e.*, for both and . This choice allows us to easily take the limit as by simply replacing by :

(3.39) |

Replacing by and converting to -transform notation instead of Fourier transform notation , with , yields the final result.

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Differentiation Theorem Dual

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Stretch/Repeat (Scaling) Theorem