This is also why wide bandwidth tube amplifiers are necessary for accurately reproducing sound:

1) they can pass through the higher order harmonics of the fundamental frequency
2) and they maintain the phase and amplitude of those harmonics

And it is this harmonic structure that makes music sound more life like.

The comment in the previous post makes this point too:

"I remember dCS had a white paper on their website some time back, that explained what [color=#0000ff]extreme band limiting in the digital domain did, and they showed it caused phase and amplitude shift, they had snapshots of it on oscilloscopes. This is known in physics as the Gibbs Phenomenon. It is never explained by the apologists of just sampling at nyquist. Any filtering you do will cause a ripple effect in the audio much like throwing a rock in a pond. Humans can hear the harmonics of fundamental frequencies way above 20k and the phase and ringing artifacts smear in the time domain. All of these things are important. If high order harmonics were not important then, for example, the type of piano or guitar you use and how it resonates would be inconsequential because they are all generating the same fundamental frequencies, but as any musician will tell you, it does matter. Why? due to the distribution of how those frequencies sit, and all of this information happens in the upper end of the frequency range.

Personally I have heard the effects of high sample rate and DSD, but it does require a controlled set up, actually I found that tube gear demonstrates it the most due to it's high slew rate, and when I managed a high end store, we had some tube preamps that could go to 100k and high sample rate was very audible on those especially DSD. I also had a CD player that had a band limit of 80k and could select the CD layer at 16/44.1k and the SACD layer, I was even able to demonstrate to audiophiles who didn't believe in DSD that it made a difference it was very audible on a non band limited player.

In short I think the people who maintain there is no difference either have never heard live music or lack the equipment and discernment to hear the difference, because the changes are very audible even to the un-initiated on the right system."[/color]

orangecrush wrote on 03/24/18 at 17:53:45:


Looks like a great DAC, let us know what you think if it enters your system.

I don't know about  some of these conclusions. In my system dynamics are almost off the hook, so I'm not sure my PS Audio DSD and its conversion of all sources to DSD64 is restricting anything but there certainly are other factors in my system (ZTPRE, ZBIT, ZROCK2).

Well, I hope that your journey takes you where you want to go with a source that doesn't need upgrading. it's interesting that my DSD gets it's OS upgraded and each upgrade is an improvement on a sound I was very happy with, so I don't see a danger to upgrades!

And also interesting that I was so adamant about keeping things as simple as possible, but adding my "Z Amigos" has both given me MORE transparency and LESS noise. And neither would matter if they weren't giving me BETTER sound. So I'm on the final leg of my different journey. . . the Anniversary Monoblocks should ice the cake for me and I'll be set for some time  (my guess).

Just thinking more about my less then stellar experiences with the ZStage and Zbox.

I think he biggest difference is when you are using very high efficiency speakers. My speakers are 101 db-spl 1w, 1m. My room is very quiet, and my power and cabling helps achieve a very low noise floor. I hear absolutely no noise or hiss even with my ear right against the speaker. The addition of the Stage and Zbox added noise and hiss which was not acceptable for me.

But, I don't think you would notice that much on a 92 db-spl speaker. But is you did hear a tiny increase, it would be much more on my speakers.

I've been trying to think of a good analogy to describe how filtering can impact sound and I keep coming back to water.  Steve often uses water analogies to explain complex audio concepts -which always creates a clear picture in my mind.  So here it goes...

Imagine an olympic swimming pool 50 meters long and 10 lanes wide.  Let's say lane 1 is the audio signal for humans, 20Hz to 24kHz to keep it simple, lane 2 (48kHz) is for dog lovers, lane 3 (96kHz) is for cat lovers, lane 4 (192kHz) is for dolphins, lane 5 (384kHz) is for robots, and lanes 6-10 (768kHz to 12,288kHz) for UFO landings, outer-space, dark matter, etc.  Your noise floor would be determined by how calm the water is, so ideally the pool would look like glass and you would be able to clearly see your undistorted reflection.

Now for your audio system design (Source/Amp/Cables/Speakers) the lane dividers would be the filters.  So you have to decide where you want to put the lane divider (and for this analogy the lane divider will be yellow rubber ducks strung together by steel cable).  So for a low bandwidth design we would put the lane divider at lane 2, and for a high bandwidth design we would put the lane divider out past lane 5.

The 50 meter length of each lane would be your audio system signal path from the source (the start line) all the way out through your speakers to your ears (the finish line at meter 50).  For this analogy we'll assume the audio source is perfect -but really we should consider, as part of the signal path, how the audio signal source was captured, recorded and mastered too.

Now we need to pass our audio signal through lane 1 of the pool.  Let's use the familiar pebbles in a pond analogy for the audio signal.  So the goal for the audio system would be to only have pebbles as the source of the ripples in the water.  So in our low bandwidth design, the lane 2 divider would have strong feedback ripples bouncing off the rubber ducks back onto our source ripples.  In the high bandwidth design the source ripples would have to cross over 5 lanes before they bounced of the rubber ducks and then have to travel back another 5 lanes to interfere with the source ripples.

We would want straight lane dividers too, so our lane dividers (filters) would all need to maintain the same bandwidth from start to finish.  For example, if one component of your audio system compressed the bandwidth down to lane 2 at any point along the 50 meter length of the lane then your audio signal would once again have strong interference because the feedback ripples coming of the rubber ducks would be much closer to your lane 1 source ripples.