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4 minutes ago, The Computer Audiophile said:

If we are to reproduce audio as close to the original recording as possible, then we can't really rely on what we think something sounds like in real life unless we were at the recording and we know what was done to it for the final release. 

 

Interestingly (and particularly with Miska's involvement here) it is odd that so many CA members upsample to DSD. As the number of recording and mastering engineers, artists and whoever, listening to final mixes via dsd, must approximate to zero, then any files so upsampled and presumably sounding different (adherents obviously claiming "better) must by definition be less accurate than the pcm original. Might as well just throw a valve in somewhere 😊

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2 minutes ago, The Computer Audiophile said:

 

I think there is a misunderstanding. By upsampling and converting to DSD people are making the reproduction more accurate in their systems. It has nothing to do with the format heard while creating the music. We take what was created and remove issues by upsampling or converting. Most DACs do this internally, so it's like the creators heard this as well. 

 

No. More accurate to what?

 

The creators will have been in a studio listening to pcm. If upsampling to dsd alters the sound (and if not why do it) it is by definition less accurate.

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7 minutes ago, The Computer Audiophile said:

 

From a measurement standpoint. If you can remove distortion caused by your DAC by upsampling, then it's more accurate to the recording. 

 

 

 

Chris.

 

Not sure if you're being a bit thick here or wilfully misunderstanding?

We're talking of upsampling specifically to dsd. 

 

The mix in the studio which was signed off was not upsampled to dsd.

Your appeal is that upsampling to dsd removes distortion.

But this "distortion" just like any other effect is inherent in that which has been signed off.

If you remove it you are no longer hearing what was heard (and signed off) in the studio

 

(I won't go quite so far as to write QED)

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4 minutes ago, The Computer Audiophile said:

 

I'm telling you it's absurd to think you can know the accuracy of anything. People need to be OK with not knowing things in life. 

 

I think I could have a fair guess as to both the position and momentum of your brain cells just now...!

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1 minute ago, The Computer Audiophile said:

 

One other point, your DAC internally either upsamples to high rate PCM or SDM. Do you turn this off, thinking it will be closer to what was heard in the studio? This is implemented because it reduces inaccuracies in the reproducing of the recording. It gets you closer as close as possible with one's equipment. 

 

I have a Gumby so no option.

 

Perhaps it's a philosophical point. Do you suppose that if Hendrix were gifted a cleaned up, remastered, digitised version of Little Wing, upsampled to dsd played back via class d amps and Magicos, he'd go "Wow!"

I somehow think not.

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5 minutes ago, The Computer Audiophile said:

 

Perhaps I'm not explaining this correctly.

 

Your Gumby internally uses Mike's super burrito filter and oversamples to the rate Mike likes. This is likely his attempt to make the music more accurate to what's on the recording. It has nothing to do with being more accurate THAN the recording. It's about not adding anything to the recording. That's why DSP is used in all DACs. 

 

 

 

 

Of course I know this.

 

Perhaps I'm not explaining correctly.

Hendrix (or whoever) was in his studio with his mastering engineer. What they heard coming out of the speakers was the sound they wanted. This is already a once (or more) remove from "the recording".

How do you reproduce that? Maybe high end is all just bollocks after all (Acid, weed, whiskey may help!)

 

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8 minutes ago, The Computer Audiophile said:

 

Totally. The sound captured and released on the recording is only a facsimile of the real thing. However, we have to use what's delivered to us. We don't know what Little Wing sounded like in the studio that day. Or perhaps, Hendrix was satisfied with the sound on the album after mixing and said this is what I want to deliver, given that I can't deliver myself to everyone's home. Thus, by converting to DSD on some audio systems we get as close as we can to the sound of the delivered recording. If we don't convert to DSD on some systems, the sound is farther away from what was OK'd for the commercial release. 

 

You're still not getting it.

No studios convert to DSD*

The commercial release was mixed and ok'd via pcm chain

Converting to DSD alters the sound (or why do it). 

Thus, by converting to DSD on some audio systems we get further from the sound of the delivered recording

 

If we don't convert to DSD on some systems, the sound is precisely what was OK'd for the commercial release. 

 

Geddit? (what are the chances you read Private Eye)

 

(*maybe there is one?)

 

 

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6 minutes ago, Samuel T Cogley said:

 

I just think it's bad form to trash another audio forum.  And that's what we're doing here, but there's enough off topic posts in the thread to make it seem like that's perhaps not the case.  But delete all those off topic posts, and this thread is just a way to trash ASR in particular and "objectivists" in general.

 

 

 

Yeah but we're all on the spectrum...

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In reverse order...

 

4 hours ago, The Computer Audiophile said:

P.S. @Andyman can you find someone on Earth who agrees with your logic on this? I'm all ears, but I think it's faulty. I'll even listen to the most objective people on this one. What does @pkane2001 say? I'm all ears. 

 

My logic is usually pretty sound (?!) However I fear you are conflating two subtly different arguments.

 

1) Whether playback via DSD upsampling is (inherently) superior (to plain pcm playback (upsampled to higher rate pcm or not)) 

 

Here the jury is still out but I raise you Mans (if he's still around - probably been banned for being a bit of a wanker at times but can be funny and knows a fair bit). Moffat too (two?) Probably mad Peter also but not entirely sure.

 

2) Whether playback via DSD upsampling provides a more accurate representation of what the artist (producer/engineer) intended

 

No it doesn't. This was my original argument (which I rather pompously labelled a philosophical point), so presumably where you (dare to) challenge my logic, and to which I still adhere (and is (surprisingly?) independent of 1) above)

 

It actually doesn't matter here, whether playback via DSD is or is not inherently superior (to plain pcm playback). It is sufficient that they are different.

Let me try it this way (rather than asdf100 or whatever he's called just repeating, copying and pasting over and over and over...)

 

For the sake of argument, let's agree that a playback chain which upsamples to DSD sounds different to one which does not.

 

Following months of toil and angst, an artist/producer/mastering engineer in the studio is preparing the final mix. Naturally there are no DSD dacs around here. The mix is perfected, being replayed via the ubiquitous pcm dac and signed off for release - phew, job done.

Now consider Chris C of this parish just happened by that day and inserted his upsampling to DSD gubbins (computer/dac - maybe even HQP 😳) into the mastering replay chain. We are agreed that the sound is now different (we could say we've applied a "DSD function")? However, if it was perfect before and is now different, it can no longer be perfect. It is by definition inferior - sadly disappointing (as mentioned earlier, it matters not one jot which replay chain is inherently "superior", merely that they differ. It may well be that the incredibly clever engineer could remaster the file so that replay via this latter DSD chain sounds identical to that of the former pcm chain. But that's of no relevance here).

The file is now released to the masses. Playback via my Gumby (upsampled to pcm384 or whatever probably similar to that in the mastering playback chain) sounds superlative.

Unfortunately Chris is still upsampling to DSD so what he hears has the aforementioned "DSD function" applied, thus detracting from perfection. (Of course had the cleverly remastered version been released, the roles would have been reversed - but they weren't). 

 

4 hours ago, The Computer Audiophile said:

 

 

Here's is where I believe you don't understand how digital audio works. 

 

All digital audio goes through DSP to get more accurate to the recording. Converting to DSD doesn't alter the sound away from the recording, it alters it closer to the recording. It doesn't matter what signal chain was used in the recording process, what matters is taking the delivered albums and playing it as perfectly as possible. 

 

For example, if you play PCM at 44.1 on a specific DAC, you'l get nonlinearities and distortions. These are added by the DAC and not in the recording. If you convert to DSD on that same DAC, you remove these nonlinearities and distortions through DSP and you can hear as close as possible what's on the delivered recording. 

 

By using DSP and filtering you get precisely what's on the recording. All DAcs work this way. It doesn't matter the process they use, PCM, DSD, R2R, Ring DAC, Sabre DAC, etc... it's about the final analog output. Using DSP and DSD on some DACs gives you more accurate analog output that enables you to reproduce the recording delivered to consumers, more accurately. 

 

 

 

How very condescending of you 😊. But it seems more likely you misunderstand. From above:-

 

Converting to DSD doesn't alter the sound away from the recording, it alters it closer to the recording

 

...so converting to DSD alters "it" (the sound?) closer to the recording. Presumably you understand the meaning of oxymoron?

 

And then:-

 

if you play PCM at 44.1 on a specific DAC, you'l get nonlinearities and distortions. [Maybe your talking here about NOS dac with no prior software upsampling? - hardly the norm]These are added by the DAC and not in the recording. If you convert to DSD on that same DAC, [errm - can't?!] you remove these nonlinearities and distortions through DSP and you can hear as close as possible what's on the delivered recording. 

 

C'mon chris admit it. You've had a chat with Miska, not really grasped what he's told you, and this is your unfortunate interpretation 😪!

 

Nevermind and no hard feelings.

 

Love,

 

Andy

 

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11 minutes ago, The Computer Audiophile said:

 

 

Andy, let's take HQP out of this and use only the DSP that goes on in every DAC. 

 

What's the purpose of this DSP? To remove things like noise, nonlinearities, and distortions among others. 

 

How is this not increasing accuracy? 

 

Presumably the intention of all DSP is to improve something. But excepting those algorithms which preclude bit perfection, in what way do they differ with respect to accuracy? Mine's more perfect than yours?

 

No need to take HQP out of this. I have no doubt Miska knows what he's doing and why.

 

 

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18 minutes ago, The Computer Audiophile said:

 

HI Andy, it will probably be easiest to digest this little pieces at a time, at least for me. 

 

I'm listening to a DAC that enables external DSP processing. When it's used, the DAC measures much better and is more linear. This provides a more accurate analog output through DSP. Removing nonlinearities is a good thing. It isn't a matter of mine is more perfect because the built-in filters can be bettered. No filter is perfect. 

 

 

 

 

 

Bringing HQP back into the conversation (it is the paradigm for such discussions and presumably what you are using?)

I occasionally glance at his software thread and all anyone ever asks is "which filter should I use?"

I'l bet you've measured more than one filter. Would be interesting to know if you have noticed a correlation purely between linearity and perceived SQ? And what else, if anything, have you measured that you feel might be notable...

 

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