When Audacity Down-Samples a Track

In This Posting, the reader may have seen me struggle to interpret, what the application ‘QTractor‘ actually does, when told to re-sample a 44.1 kHz audio clip, into a 48 kHz audio clip. The conclusion I reached was that at maximum, the source track can be over-sampled 4x, after which the maximum frequencies are also much lower than the Nyquist Frequency, so that if a Polynomial Filter is applied to pick out points sampled at 48 kHz, minimum distortion will take place.

If the subject is instead, how the application ‘Audacity‘ down-samples a 48 kHz clip into a 44.1 kHz clip, the problem is not the same. Because the Nyquist Frequency of the target sample-rate is then lower than that of the source, it follows that frequencies belong to the source, which will be too high for that. And so an explicit attempt must be made to get rid of those frequency components.

The reason Audacity is capable of that, is the fact that a part of its framework causes a Fourier Transform to be computed for each track, with which that track is also subdivided into overlapping sampling windows. The necessary manipulation can also be performed on the Fourier Transform, which can then be inverted and merged back into a resulting track in the time-domain.

So for Audacity just to remove certain frequency ranges, before actually re-sampling the track, is trivial.

If my assumption is, that QTractor does not have this as part of its framework, then perhaps it would be best for this application only to offer to re-sample from 44.1 kHz to 48 kHz, and not the other way around…

Dirk

 

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