Originally Posted by stv014
Null testing/difference extraction is very sensitive to frequency response and delay differences. That is why audio recorded from actual analog DAC outputs, which do not have perfectly flat response, and also have small random delay variations due to the clock frequency not being entirely constant (basically, very low frequency jitter), will produce a relatively high difference signal. Notice that the sound card recordings all have a notch in the difference signal at the 1500-2000 Hz range, because that is what the frequency of the tones used for matching the levels and delay was.
In the case of D.wav, the minimum phase filter has a group delay that is zero at DC but increases towards the Nyquist frequency, and therefore the difference is also greater in the high frequency range. This file also has the highest group delay in the treble range.
I was wondering about picking up additional signal changes due to the Xonar's playback out the analog line--I take it the analyses posted on page 7 are consistent with that. It makes the desired outcome--comparing 44.1 to 192 by comparing encodings of the analog playbacks of both, both captured at 192/24--inherently problematic. By removing one variable, sample rates at playback, more variables from the playback card (here, Xonar) are introduced.
From my post #46:
"On the one hand, this removes the effects of any difference purely on my end in DAC handling of differing playback rates at playback time--because both in this test are in 192_24.
On the other hand, it adds two new factors---not that they are known to have effects, just that these are differing experimental conditions:
the DAC processing on stv014's device to send the 44.1 file to analog playback,
the effect of A/D on stv014's 192-encoding capture."
I'm persuaded by all the above that the fine 'tuning' of the available parameters in resampling is important. Subtle, of course, but the differences are there.
Edited by UltMusicSnob - 8/24/13 at 2:33pm