QUOTE
However, to achieve a higher degree of fidelity to the live analog reference, we need to convert audio using high sampling rate even when we do not use microphones and loudspeakers having bandwidth extended far beyond 20 kHz. Listeners judge high sampling conversion as sounding more like the analog reference when listening to standard audio bandwidth.
and this quote struck me as very, very odd.
(read the article first, or at least the conclusions/test setup bits before posting.. the phrasing seems to be a bit awkward in places, but i can't really help that)
because unless i'm really, really reading this wrong they're saying that if you upsample to the highest SR your sound card can take before playing it back - no matter the quality of the input material - it will be perceived as 'more natural'.
(mind you, i don't really see why you'd need to do what they suggest next, namely:
QUOTE
These results suggest that the archiving community should consider using high-sampling conversion to ensure transparency even if the recording is made with standard audio-bandwidth transducers, and when digitizing older recordings made with bandwidth-limited analog systems.
just turning on SSRC in your foobar dsp chain would seem to be enough)the point, really, is that i almost can't believe that i'm understanding this right.
A second point: even if they are correct, it would seem that their conclusion that anything >20kHz is a waste of bw is somewhat hasty, and they imo should've added a 'middle' test - to say 40kHz - just to narrow down where the added bandwidth became overkill (more).
