Originally posted by niktheblak So what if the signal would have been recorded at say 96 kHz in the first place, with an analog lowpass stop-band of at say 92-96 kHz. In this case, the resulting 96 kHz signal could be digitally lowpassed (much narrower stop-band) at 22.05 kHz and decimated with a factor of 0.5 leaving us with full, usable 22.05 kHz frequency band.
Originally posted by 2Bdecided I don't agree with the opinions written here.Almost all ADCs (and DACs) use oversampling, which means that the digital signal is sampled at many times the target sample
domain ringing of the filter). However, some systems contain fantastic filters, which get you almost all the way up to 22kHz without any aliasing problems. It is possible to generate a filter which passes 22kHz perfectly, and attenuates 22.05kHz by 120dB - you can even do it in Cool Edit Pro!
Unfortunately, the anti-alias digital filters aren't always great because of cost constraints, but they can be very good, giving almost textbook response up to 22kHz. The phase response can be perfectly linear, though again, cost contraints can compromise this.
But to assume that there's never anything useful up there would be false. If you record at 88.2kHz and resample optimally, you can be sure that what's up there is accurate.