
regebro
Insane Genious!
I had a revelation a couple of days ago. I got a theory why recording in 48kHz and the downsampling to 44.1 might actually be better. One reason was mentioned before, that the resolution is higher so that digital effects, primarily reverbs, could work at a higher quality. But that difference is only a couple of percent, so I scratched my head and thought that it didn't feel right.
But here one big difference: The anti-aliasing filters.
When you convert from analog to digital you need to filter away all sounds above half the sampling frequency to avoid aliasing. That frequency is 22.05kHz and 24kHz respectively.
To preserve the sounds at 20kHz, that means that you have to have a very steep filter to block out everything at 22kHz. Steep filters like that are hard to make, and you get phasing problems, and stuff. But if you have 48kHz sampling frequency, you have to block things at 24kHz, so obviously, the filter you use only have to be half as steep to do the same job. So, here is a difference between 44.1 and 48 where the practical difference *doubles* between formats. No skimpy few percents, the filter actually has to be only half as steep in 44.1 and in 48. And of course, if you compare with 96, the filter now only has to have a twelfth of the slope of a 44.1 kHz filter.
In theory you'd like a filter that has 0dB damping at 20kHz and 96dB at 22kHz. But in practice you will cut corners. You'll either start the cutoff before 20kHz, or you will not have full damping at 22kHz. None of these are any real problems, since most people doesn't hear the highest frequencies, and you'll never have a 0dB 22kHz tone anyway.
But with 48kHz, you don't have to cut corners as much.
I suspect that most sound cards would use the same filter in both cases, actually, either raising the cutoff-frequency by 2kHz when doing 48 kHz, or have a cutoff of maybe 30dB at 22kHz and 60dB at 24kHz, which with high frequency sounds could cause some minute aliasing.
The slight loss in HF content, or the minute nearly inaudible aliasing might be enough to make the sound be percievably worse, especielly when processed.
But as I said, this is only theory. What do you think? Sounds reasonable?
But here one big difference: The anti-aliasing filters.
When you convert from analog to digital you need to filter away all sounds above half the sampling frequency to avoid aliasing. That frequency is 22.05kHz and 24kHz respectively.
To preserve the sounds at 20kHz, that means that you have to have a very steep filter to block out everything at 22kHz. Steep filters like that are hard to make, and you get phasing problems, and stuff. But if you have 48kHz sampling frequency, you have to block things at 24kHz, so obviously, the filter you use only have to be half as steep to do the same job. So, here is a difference between 44.1 and 48 where the practical difference *doubles* between formats. No skimpy few percents, the filter actually has to be only half as steep in 44.1 and in 48. And of course, if you compare with 96, the filter now only has to have a twelfth of the slope of a 44.1 kHz filter.
In theory you'd like a filter that has 0dB damping at 20kHz and 96dB at 22kHz. But in practice you will cut corners. You'll either start the cutoff before 20kHz, or you will not have full damping at 22kHz. None of these are any real problems, since most people doesn't hear the highest frequencies, and you'll never have a 0dB 22kHz tone anyway.
But with 48kHz, you don't have to cut corners as much.
I suspect that most sound cards would use the same filter in both cases, actually, either raising the cutoff-frequency by 2kHz when doing 48 kHz, or have a cutoff of maybe 30dB at 22kHz and 60dB at 24kHz, which with high frequency sounds could cause some minute aliasing.
The slight loss in HF content, or the minute nearly inaudible aliasing might be enough to make the sound be percievably worse, especielly when processed.
But as I said, this is only theory. What do you think? Sounds reasonable?