Are high sample rates making your music sound worse?

Wow, this is getting out of hand in breadth and how fast it progresses. Someone commented on what i said (quoted me) and then went on to speak of Chord DACs having the best filtering they knew of. It **appeared** that they were using it to counter my argument that up/over sampling helps with the filtering.
Let me be clear:  Chord over/up samples.  Their data stream runs at 104 mHz/mbps - far, far, far above the raw output of redbook ( around 1.5 mbps).
Their sound is not all due to filtering either.  They pay special attention to timing and jitter (as one must, since the reconstruction filter integrates over the sample value and the time).
They further use a different method to get the analog pulses (flip-flops, let's not go down that art-hole) that has some claimed advantages.
But it in NO WAY counters what I wrote.
But with the loose writing here it difficult to even understand what point people are trying to make.
G

@itsjustme 
 
I know the Chord upsamples, so does the Benchmark, and so do many others, I’m specifically referring to upsampling before the DAC. DACs do this as it helps with jitter, doing this before the DAC doesn’t help.

" know the Chord upsamples, so does the Benchmark, and so do many others, I’m specifically referring to upsampling before the DAC. DACs do this as it helps with jitter, doing this before the DAC doesn’t help."
No, they do it to interpolate and reduce the challenge for the reconstruction filter.  Again, don't believe me, read a Burr-Brown application note if you prefer.
There was also a point about up sampling being no longer bit perfect.  If you understand interpolation, which is a distinction without a difference. If you understand interpolation, you know that it creates new data points which a) raise the frequency of the noise to be filtered and b) invent a new intermediate level - beyond 16 bits -- which in effect raises the resolution.  It is absolutely true that no new data exists and it is absolutely true that in a world of perfect, phase-coherent, 200 dB/octave filters, it would be un-necessary. But we are not in that world. So, at 88.2 kHz we get effectively 17 bits.  at 176.whatever we get 18 bits.
I acknowledge that at 96 we do nto have a simple multiple, BUT -- HUGE BUT -- all the original data points remain adn can be reconstructed so in practice it is bit perfect.  Its like saying "i used to have $100, this crook gave me $2 and now I don't have $100 anymore".  Uh, true, you have $102.  You are free to throw $2 on the ground - and the DAC is free to toss any bits on the ground; but it will use them to make its later job easier.
And yes, there are differences pre-DAC and in-DAC but they are ALL BEFORE the actual conversion - either via PDM (sigma-delta) or PAM (ladder).

@itsjustme

Using a multiple to upsample is an easier job and hard to mess up. Using a non-multiple helps with jitter but is harder to implement.

As for up/over sampling before the DAC, it matters whether either side takes care of intersample overs, the Benchmark models for instance does take this into account and lowers the level before oversampling.

Interpolation does “destroy” the original data while synchronous upsampling preserves the original data. It matters how good the interpolation is. Going with your currency analogy, it would be like if I had a $100 USD but someone took it and gave me $100 in Bahamian currency, which averages a 1:1 conversion with USD, but sometimes it’s not, it has been worth 0.98:1 and sometimes 1.02:1. 

Stereophile’s measurements of DACs shows how much aliasing occurs by using a high frequency test tone, if you look at the AudioQuest DragonFly Red’s measurements for instance, the aliasing image at 25kHz is below -100dB.

I can’t comment on phase though as I haven’t seen that tested before, but we are very forgiving of phase errors (not talking phase mismatch) in the treble that I doubt any decent DAC will have an issue.