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48kHz vs 96kHz: audible?

As a so-called audiophile, it is easy to lose one’s balance within many discussions and end up doubting, or at least questioning, whether that subtlety which one hears is real or imaginary.
 
Today, while engaged in a pastime, I was playing Holst’s "The Planets" in the background, but not at a low volume. I thought that it didn’t sound right. The strings in particular sounded a little abrasive. I noticed this on "Mars," the first composition, so it didn't take me long to perk up. On closer examination, I noticed that the DAC front panel was reporting 48kHz sample rate. I knew that this version of The Planets is 96kHz. Sure enough, JRiver Media Center (MC) was converting all PCM data (whether higher or lower) to 48kHz upon playback. I fixed the MC settings back so that all PCM rates play back at their native rates (up to the capability of my DAC), and all is well now.
 
Sometime in the recent past, whether due to an application or OS upgrade (of which there was one a few days ago), the MC sample rate conversion table got corrupted or reverted to a default configuration.
 
It would seem that I am able to hear the difference between 48kHz and 96kHz, at least under these circumstances. The difference was enough that I noticed it while passively listening (I was focused on drawing; the music was “background”) before I suspected a technical issue.

I wonder whether I could have heard this difference in a formal ABX test session? From my past experience with ABX testing, when the differences between the test objects are subtle, observations could easily have been obfuscated due to mental noise consisting of test anxiety, listening fatigue (to same passage over and over) and tedium. Whereas, in my case above, I noticed the difference when I was relaxed and focusing on something else entirely.

I am interested in thoughtful replies.

128x128mcdonalk
yoyoyaya's avatar
yoyoyaya
609 posts
 

This whole area is a bit of a minefield. The ideal way to test this hypothesis would be to take a piece of music that can be mechanically performed -using something like the technology that was used for the Zenph reperformance of the Goldberg Variations. One would then have the computerised piano play the piece of music twice and record it at the two sampling rates/word lengths, and then double blind test it across a stratified sample of listeners using a high resolution playback system.

In other situations where upsampling and downsampling is involved, it's hard to discern cause and effect.

pjb255livecom
28 posts
 

Qobuz is my sole front end, and I hear a slightly more refined attack on each note on most hi-res selections. However I find hi-res content overall of inconsistent quality and less the driver of SQ (compared to Redbook) than the recording/mastering put into it, so much so that I no longer even care what its published resolution is, FWIW.

zgas-music
78 posts
 

Hmm. I suspect that what you were hearing may not be SQ difference between sample size, but problems in how your box downsamples digital signals. 
 

Downsampling problems show up in edge cases, like high violins, reverb tails, etc. For downsampling to sound good, dither (digital noise) needs to be used to minimize aliasing/rounding errors during the conversion. There are several ways to do this, which is best done when mastering the recording. Sounds like your streamer might just be lopping off bits on the fly?  And you heard it.  Might be interesting to find out how your box downsamples and what kind of dither is used, if any. 

A lot of problems can be introduced when sample sizes get changed without close attention. 

oldaudiophile
283 posts
 

@mcdonalk , just a quick question for you: You say you were listening "in the background" while engaged in some other pastime. Once you noticed or sensed the sonic anomaly, did you, then, complete your final evaluation by doing some bona fide critical listening from the sweet spot? I can only assume that you did so in your final analysis because no matter how great your speakers' off-axis response might be, the best position for critical listening is the sweet spot.

johnss
564 posts
 

yes, significant difference between 48k and 96k, has already been done in studios quite a few times. 

next level up is 192k, which is slightly better than 96k, but at these 192k sampling rates and higher, you need a master clock to sync the data. 

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