Dreadhead

I got my wife this print of a piece called critics that was a paintbrush filled with goblins, thought it was suitable.

I think that may be part of the problem. Honestly how is an ES-2 worth that much? Maybe Mikhail figures if he can charge what he wants then he can do whatever he wants too.

This whole thing is mind blowing. How he got this far in this business is beyond me.

Ok then send it to me

Wait Mikhail lies? .... I hope this works out Icarium

At least we're in agreement that it's the mix/playback mechanism, not that vinyl is superior. As far as your sample rate conundrum I can offer no advice as it does not at all match my experience. Maybe the digital or analog filter is different between the two sample rates in the DAC. I'm can only guess. If the sample rate conversion and bit depth truncation is done well then there should be no audible difference. I have a few masters that were done at 88.2 and 96 and I can't find a difference with them when I down sample either.

It's in the mix my friend it's in the mix. Hence the popularity of needle drops. If digital is responsible for the issues with how it sounds then why do many people (including you) love the digital representation of the analog recording that was made after playback? Digital just gives you back what it's given (if it does it right).

LOL. It always takes more money to hear what people want you to hear. That's my favorite part of this hobby. I will never ever own either of those and it's not a matter of money in either case. I have measured R10s and while they are fun (lovely even for classical) they are not detailed or transparent. K1000s just don't really fit me well thought I did enjoy them out of a 16gain balanced beta22 using my DAC3 as a source. Scratch that I may buy k1000s sometime.

Spacial positioning/sound stage is different for headphones I fully agree. That said I believe that there is nothing like them for judging the differences between two mixes (unless we're talking sound stage). I don't know how many times I've suddenly learned a track that I loved and thought was wonderfully recorded is utter crap once I listened to it on phones. Luckily in most cases I can still enjoy the music though. Anyway that's not the point I was trying to get across so I will drop it now.

Are you saying that speakers present more information than headphones? I'll have to heartily disagree. For the record: Bel Canto DAC3 (measured noise floor below 110 dB) Headamp GS-X HD650s,SA5000s,D5000s and Edition 9s. (I think we were mostly using the last 2 when he was over but I've done it by myself with the others)

I concur. People have finally started to take the data and rebuffer/reclock/resample it so you aren't dependent on the disc. That's the way it should be

Dusty I have done it. I took a digital track and set re-sampling and bit depths all over the range (44.1-96/ 16-14) and I or the other person couldn't hear any difference. This was using the SRC sinc interpolator (which is a brick wall filter btw). The reason you add dither is to make sure that the noise floor stays wide banded as you point out. I had oversimplified.

I didn't say worse I said different, the highs were higher after the sample rate switch if I remember right. If you're charging someone to have 24/96 it's always better if you can hear a clear difference which you can if they change the filter. Call it optimized or whatever you want. It shouldn't be there.

So do I! As I said in the DAC64 case. We switched sample rates and bit depths and the sound changed then did the same with my DAC3 and couldn't hear any difference. As I said the owner then sold it and it fell off my list of dacs to consider.

Much like some SACD's have a different mix on the CD layer than on the SACD layer. There is a difference but it's not the sample rate (if done well) or bit depth.

We are a low pass filter (with weird response and varying effectiveness). A low pass filter does not care what it's blocking out it just blocks it out. A lot of dacs have a brick wall filter on them because you don't want high frequency noise output into op-amps that have problems with them. Sample rate down-sampling has to be done exceedingly carefully but most just use a higher order interpolation and then a brick wall to prevent the noise leakage and of course noise shaping or dither.

Well then people were doing the bit depth conversion wrong (maybe the normalization?) because the noise floor on the LP is well above anything in that the 24to16 bit truncation should see.

True enough I didn't discuss noise shaping to prevent the harmoinc rounding but as Grawk pointed out it's on the inaudible LSB.

Yup. When stuff is below the noise floor it's below the noise floor and that's it.

Or because some DACs do weird things to make the two sound different. The chord DAC64 is a good example of this (at least to my friends ears since he sold it afterward). Truncation from 24 to 16 should be like this (just an example): 24bit: .9230232332102301239123091233120931209 16bit: .923023233210230123912309 Sample rate conversion on the other hand can be a bit of a problem and there are different theories on how to do this that have their own problems. In the end each is an interpolation. There have been tests done where people have taken a 24/96 stream and created a DBT box that either left it alone or resampled and trunctated it (correctly) and no one could tell the difference. Actually if I remember correctly someone could tell but he admitted it was because the "click" when the stream engaged was slightly different not anything to do with the music.

I wish I could come

so true. Most sound guys deserve to be shot. If you're listening to unamplified jazz (other than vocals) or classical then argument doesn't hold much water though.

I absolutely agree it will reduce your DNR SNR but so does a regular volume control so I don't see that as much of an issue. Also you are now asking your output stage to amplify a smaller signal and hence you should get a more linear response out of it and lower THD. It's a trade off between better linearity in the DAC portion but worse in the output stage. I think at best it's a toss up between attenuators and digital volume control. You probably only need the top 16 bits of data at max volume and people can't hear below whatever they can't hear below so when the data at the bottom drops out it is just gone.

They think that the 32 bit is now covering the smaller range but that's not how it works as we both know.

As stated for a lot of 24 bit dacs the data is still there too. With the 32bit dac the data is swallowed in noise. My DEQ does it's math in 32 bit and then outputs it back out in 24bit that doesn't mean a damn thing other than the math is accurate inn 24bits. The Bel Canto does digital volume control in 26bits for the same reason but to be honest the last two bits beyond 22 do nothing.