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The analog source is never perfectly limited to 20 kHz because very steep filters are expensive and they may also degrade the signal in other ways, because their transient response is not completely constrained by their amplitude-frequency characteristic.

This is especially true for older recordings, because for most newer recordings the analog filters are much less steep, but this is compensated by using a much higher sampling frequency than needed for the audio bandwidth, followed by digital filters, where it is much easier to obtain a steep characteristic without distorting the signal.

Therefore, normally it is much safer to upsample a 44.1 kHz signal to 48 kHz, than to downsample 48 kHz to 44.1 kHz, because in the latter case the source signal may have components above 22 kHz that have not been filtered enough before sampling (because the higher sampling frequency had allowed the use of cheaper filters) and which will become aliased to audible frequencies after downsampling.

Fortunately, you almost always want to upsample 44.1 kHz to 48 kHz, not the reverse, and this should always be safe, even when you do not know how the original analog signal had been processed.



yeah but you can record it in 96kHz, then resample it perfectly to 44.1 (hell, even just 40) in digital domain, then resample it back to 48kHz before sending to DAC


True.

If you have such a source sampled at a frequency high enough above the audio range, then through a combination of digital filtering and resampling you can obtain pretty much any desired output sampling frequency.


the point is that when down sampling from 48 to 44.1 you can for "free" do the filtering since the down sampling is being done digitally with an fft anyway




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