Editing Interpolation (section)

==In digital signal processing==
In the domain of digital signal processing, the term interpolation refers to the process of converting a sampled digital signal (such as a sampled audio signal) to that of a higher sampling rate ([[Upsampling]]) using various digital filtering techniques (for example, convolution with a frequency-limited impulse signal). In this application there is a specific requirement that the harmonic content of the original signal be preserved without creating aliased harmonic content of the original signal above the original [[Nyquist frequency|Nyquist limit]] of the signal (that is, above fs/2 of the original signal sample rate). An early and fairly elementary discussion on this subject can be found in Rabiner and Crochiere's book ''Multirate Digital Signal Processing''.<ref>{{Cite book |title=R.E. Crochiere and L.R. Rabiner. (1983). Multirate Digital Signal Processing. Englewood Cliffs, NJ: Prentice–Hall. |isbn=0136051626 |last1=Crochiere |first1=Ronald E. |last2=Rabiner |first2=Lawrence R. |date=1983 |publisher=Prentice-Hall }}</ref>