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[kmmcd]

Several days ago when working at my drafting table, I was playing Holst’s "The Planets" in the background, but not at a low volume. I was using Media Center to play this file to my DAC via USB. I thought that it didn’t sound right. The strings in particular sounded a little abrasive. I noticed this almost right away 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, Media Center 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), or maybe even an error on my part, 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 primarily focused on working at the drafting table at the primary listening spot; 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 first raised this experience on the Audiogon forum several days ago, and multiple replies suggested that Media Center was not performing the downconversion correctly, resulting in audible artifacts. I have been using MC for many years and have come to respect its DSP features, but unless I raised this possibility here, I would not be being thorough or open minded about this.

Any insights on why this unintentional downconversion would be noticeable, particularly when I was not looking for it, are welcome.

If you are interested, here is the thread on Audiogon:

https://forum.audiogon.com/discussions/48khz-vs-96khz-audible

[mojave]

In Tools > Options > Audio > Settings, did you have "Use Sox for resampling" checked? It should be. There can possibly be down sampling artifacts without it turned on.

Wavelengths get smaller and smaller as they get higher in frequency. Sample rate is related to the smallest size of measurement that can be taken. A higher sample rate lets one measure smaller wavelengths.

A sample rate of 48 kHz can measure wavelengths as small as 24 kHz. A sample rate of 96 kHz can additionally
measure wavelengths between 24 kHz and 48 kHz - or one octave higher in frequency.

It is like having a tape measure in 1/16 divisions that can measure wavelengths to 24 kHz and then getting a tape measure in 1/32 divisions that can measure wavelengths as small as 24-48 kHz. For all measurements that are no smaller in size than 1/16", both tape measures work equally well. Both will measure 1" at exactly 1". Likewise, both 48 and 96 kHz will measure a 24 kHz wavelength exactly the same.

There is rarely any content above 48 kHz for several reasons:

• Few studio microphones will measure above that frequency
• Few speakers will play above that frequency
• Air absorption makes it impossible to record or playback those higher frequencies when the mic or listening position is more than a few feet away from the source
• Instruments don't have any harmonics above 48 kHz except some cymbals.

[kmmcd]

Why is enabling SoX superior to the default conversion method (SSRC, I believe)? Cursory research has revealed that JRiver at one point stated that SSRC was adequate.