Is USB straight from a Mac computer to a DAC really that bad?

[beerandmusic]

I also used to believe that higher sampling rate meant more resolution, but what i read recently is that you don't hear more music because of more sampling times, it's just that upsampling pushes distortion (due to dac's reconstruction filters) above the audible spectrum....but that you don't need dsd rates to do that.  Most all recordings are at 44.1K and that covers the entire audible spectrum.

 

Given:

a) 44.1K covers our audible hearing

b) most recordings are at 44.1K or 48K

 

My current understanding is that 96K should be more than sufficient, and why would we want to upsample higher and even mess with DSD which introduces more problems.

 

No blind test to date has supported that DSD sounds better than high rate pcm.

 

Until I have just recently revisited my thoughts (clearly not an audio engineer), but I am actually thinking PCM may be better than DSD now.

[beerandmusic]

35 minutes ago, beerandmusic said:

 

I don't want to agree with one side of the camp or the other.  I am unbiased, and just want the facts.

Two years ago, i started this a topic with this same thinking...

============================================================

As a matter of fact, here is an exact quote of mine from 2 years ago:::

 

I accept noise of all types affect SQ, but everyone kept suggesting that the DAC gets the music bits perfectly, but that noise affects the dac in processing.  This suggests that the DAC doesn't even receive the music bits with accuracy due to noise....big difference.  Some suggest that dacs can resolve for most noise, that's all dandy, but if the music is already inaccurate before it gets to the DAC input pin(s), there is no way to fix it at that point, if it is already corrupted.

===========================================================

The topic ended with basically no one agreeing with me or IFI....that bits lost are either SO RARE and even if they do happen it would cause a dropout, not a difference in SQ....but if people think differently today, that certainly goes along more with what I find logical.

 

To clarify I was quoting articles by IFI and Gordon Rankin, and everybody (at least everybody that spoke up), found exception to my statement that audio transmission is not bit perfect....so i accepted the consensus...and now it appears that there are people that may actually believe what I was trying to say 2 years ago....well we will see where it goes from here.

[beerandmusic]

19 minutes ago, audiobomber said:

I agree that lost bits will cause a pop or dropout. Lost bits are not the reason that digital systems sound different. You really should take ten minutes to read the article I linked, it explains jitter and its various origins extremely well. For example:

6. Power subsystem
The DC power applied to each of the devices that must process or transmit the digital audio signal is critical. If this power varies in voltage, the devices will react differently to the applied digital signals. Power "noise" as it is referred to is probably one of the largest contributors to jitter. Voltage changes or "voltage droop" can happen anywhere on a circuit board, power cabling, or even on the silicon itself. Changes in power voltage will change the speed and reaction times of digital logic that is transmitting the digital signals resulting in jitter.

8. Digital Cables
Cables don't actively add jitter to the signal, however they can slow the signal transitions or "edges". When the edges are slowed, the receiver or buffer at the cable destination is less likely to detect the transition at the correct time with certainty, which results in jitter.

10. Printed circuit board effects
There are at least two effects on a circuit board that can cause jitter, including signal crosstalk and ground-bounce. Crosstalk occurs when traces with high-speed signals are spaced closely. One signal induces voltage on the other signal. It is obvious how this can add to jitter. Ground-bounce occurs when the signal return current see a high-impedance on the circuit board due to ground-plane splits or long return paths. This creates a voltage drop in the ground-plane or return path. This voltage drop causes the signal to shift in voltage, which can result in jitter.

http://www.enjoythemusic.com/magazine/manufacture/0509/

 

none of that speaks to what the actual sound difference is, that jitter by noise causes...i vaguely remember (but i may be wrong), that someone (archimago maybe?) that shared what "jitter" sounds like, and if i recall it was like hiss.  If i hear ticks or hiss, then i would be concerned, but I do not.   my point is that i believe that "modern day dacs" should compensate for any noise on usb line, and my current belief is that they do.  Provided they have what "unsion" has or better which includes, use their own 5v reference, isolating the usb's 5v reference, galvanic isolation, and reclocking.  I am sure there are other modern day dacs that do this as well, and some are now using LPS's as well.

[beerandmusic]

12 minutes ago, The Computer Audiophile said:

In my experience jitter can round the edges of transients and make music sound soft. 

 

Ok, if i accept that, even though i believe i saw a posting at one time that actually recorded jitter inserted into a signal and it sounded like hiss, but i suppose different kinds of jitter can cause different things.

 

Let's assume it causes distortion in the audible range that actually makes music lose detail.  (I have heard loss of detail and actually have a test track that i do critical listening for it, and have heard different degrees of loss detail).

 

But it is still my contention that a well designed dac should compensate for this....and to be honest, i do believe the unison does this quite well.  They market that it provides it's own 5v reference, that it is galvanically isolated the usb, and that it reclocks.

I now can hear out my general use pc the same music of my test track that before i only heard from dsd over dlna or by playing same track in a usb slot.  (granted i have never tried usb toys before or specialized music pcs).

 

I have been griping since several years back ago, that DACs should be designed to compensate for any noise on the usb bus, I think the unison finally does that.  There may be other dacs that do it as well, but of the 15 or so i have tried, i have always fared better with dsd via dlna or using a thumbdrive, and that was 7-8 years ago now....back then i would just do dsd via dlna using a used $80 sony bluray player....i never got caught up in the usb toy thing or the more recent sotm or sonore enet products.  The unison interface playing just plain pcm is very close if not equal to the dsd via dlna i have experienced...and DLNA has it's own issues...so i am finally content with usb.

 

[beerandmusic]

9 minutes ago, davide256 said:

If you want not to be years behind in where computer audio has gone, you need to devote a weekend to reading the thread below. A lot of research by many different CA members with converging results and some key findings linked/ summarized up front. I ignored it in 2017-18 only to find out I had spent time and money duplicating results/conclusions already reached

in the thread and needed to catch up.

 

 

 

there is more than one way to skin a cat, and that thread is NOT for me (not knocking it, i am just not an audio DIY'er).

 

I was streaming DSD via DLNA via a cheapo SONY bluray player long before that thread or popularity of usb toys or music via enet existed...and I swore by it, when everyone else told me i was crazy.

 

I believe that modern day dacs have worked on usb deficiencies while everyone in that thread has found other solutions to the usb problems.

 

I believe a noisy usb can cause problems down the chain, but i don't believe it can corrupt the digital data before it is recieved by the dac (and even if it did, it would cause noticable dropouts).  Perhaps the noisy 5v or even the noise on the bits can wreak havoc on older dacs...but that a well designed dac should be able to compensate for this noise, and I believe the unison finally does that, and i am sure there are other dacs that do as well....technology does advance, and there is always more than one way to skin a cat.

 

[beerandmusic]

16 minutes ago, davide256 said:

what I've experienced is that timing degradation in the source causes background instruments to lose definition

 

This is what i always test for.  The track by Chet Baker "ALONE" is the "PERFECT" test track.

Depending on different solutions I have tried, there is a very light instrument in the right speaker, that is either clear, extremely faint, or non-existent.  I finally hear it vividly now using unison out my every day windows PC that has been absent in past, that i could always hear when streaming or via a usb thumbdrive.

 

[beerandmusic]

1 minute ago, The Computer Audiophile said:

This is a popular thing to say, mainly by objectivists, with respect to digital audio. However, I've never seen anyone put specifics to the "well designed" DAC. What is considered well designed to the level that it's immune to external issues and the cure to these issues isn't worse than the issue(s) itself?

 

I am not a dac engineer, but Schiit thinks he has something with his unison, and it is the first thing I have heard that compares to my experience streaming dsd via dlna.

 

What he markets is complete electro mechanical galvanic isolation, removing and using his own 5v ref, and reclocking.

 

I reported this on another thread at another site, and someone made the remark that all good dacs should do that and they believe that schiit is just catching up to other more highend dacs....I don't know if that is true or not, but it is the first time i have experienced what I am experiencing now...again, i have never tried any usb "toys", instead suggesting that i would wait until DAC engineers got their act together, and stick with dsd via dlna until then.

[beerandmusic]

1 minute ago, 6aardvark9 said:

 

So you're happy with what you've got... end of story

 

I don't know if it is equal to other solutions or not..i just know my test track that i have used....it's a lot better, but whether it is the end-all or not, i cannot say.

[beerandmusic]

Just now, The Computer Audiophile said:

Which Sony device are you using as DLNA endpoint that receives your DSD?

I don't have it anymore...I got rid of it when i found the marantz sr6013 is equal in dsd dlna.  I will search the threads from 5 years ago or so and see if i can find it.  It was a BDP something and was llike $100 used....but my marantz sr6013 streams dsd just fine now so i use it instead....i also like using a marantz as a preamp for purpose of audyssee and just have used different amps to drive my fronts...i will find the model and let you know....

[beerandmusic]

Just now, The Computer Audiophile said:

Ah, no worries. I thought you were still using it. I'll look at that Marantz. I like Marantz stuff and have had good success with it over the years. 

BDP S-790

 

Another thing that is nice about using the marantz sr6013 as a preamp is that it supports alexa/heos (smile)

 

I don't think there is anything more convenient...sure, i like my critical listening moments, but my wife likes that she can just say alexa, play chet baker on den, and it will turn on the amp, and start streaming from amazon prime.  Or you can tell it to play from your own playlists and use your own music.  but the one we fight over most is alexa turn up or turn down volume on den.  And it's so nice to just say "next song" or turn off den.

 

[beerandmusic]

25 minutes ago, The Computer Audiophile said:

Ah, no worries. I thought you were still using it. I'll look at that Marantz. I like Marantz stuff and have had good success with it over the years. 

I really wouldn't expect someone that can afford wilsons to use marantz stuff, but it can be a great preamp for audyssee eq and alexa compatibility....i don't know if it would degrade a levinson amp or whatever you use for your amps (probably), but it sounds great driving a mcintosh amp for fronts (smile)...it also works great for multi-channel home video (smile)....

 

btw, the sony s790 if i recall correctly could also be used as a "quiet" usb thumb drive player to play native dsd files....sounded great....a bargain and that was about 8 years ago....can be found today for $60 or so.

[beerandmusic]

9 minutes ago, The Computer Audiophile said:

Marantz makes great stuff! Another brand that makes stuff with incredible value is Parasound. 

 

I currently use Constellation Audio for amps and preamp (when needed). Given my income (I'm not wealthy by any means), If I had to pay full price for audio gear, Parasound would be my first stop. 

 

I don't think parasound makes a multi-channel av with multi-eq that supports streaming dsd and alexa (smile)...but i do own a parasound 2250...best bang for the buck for what it does (used can be found for a dance).

 

I don't think there is a better all-around product than the marantz sr6013 or current top model in same line, and you can always use it as a front end to your amp of choice....

 

as for amps, used pass labs thresholds are probably closest to mcintosh for my enjoyment that can be found at reasonable cost.

 

[beerandmusic]

27 minutes ago, davide256 said:

mm, I'd use that that as a warmup exercise but not challenging enough to be a test for me. I do like Chet though.

 

Its not a high quality recording but I  like the Jurassic Park II sound track for interesting background details... the tympani  parts are very telling on whether you are reproducing

tympani authentically.

 

I use it because i have it in various formats including DSD (it even has a 44.1K on youtube), and it is EXTREMELY easy to pick out..   I like comparing both 44.1K from youtube, playing a wav file from accurate rip, and DSD, and comparing across any solution.  It makes it very easy where i have had difficulty in past doing a/b comparisons....plus the fact i love the song, makes it easy too (grin).  If you listen to the right channel there is a very light instrument that can be heard at different degrees of vividness or even non-existent depending on how you are playing it and what hardware you are using.

 

My main desire was to be able to find a usb solution that i could be content with (without use of usb toys), as playing DSD via DLNA just has it's quirks including ability to fast forward, and sometimes just has other hiccups of different sorts, that you don't get when playing usb...plus i just wanted a solution that i could listen to from my everyday windows pc that i could be content with.  I "think" i am content with unison.

 

 

 

[beerandmusic]

15 minutes ago, davide256 said:

 

this digitally restored 1911 film probably falls along the same lines... the closer something is to what your senses normally experience, the more real it is to you

 

 

 

I just looked at your system details and it says gungnir mb unison...i thought you said you used a mojo, or am i mixing you up with someone else?

[beerandmusic]

36 minutes ago, davide256 said:

mm, I'd use that that as a warmup exercise but not challenging enough to be a test for me. I do like Chet though.

 

Its not a high quality recording but I  like the Jurassic Park II sound track for interesting background details... the tympani  parts are very telling on whether you are reproducing

tympani authentically.

 

out of curiosity, how do you play this jurrassic park ii soundtrack? from youtube like the ny scene you shared?

[beerandmusic]

1 hour ago, davide256 said:

in transition. selling Gungnir, mojo is interim solution enroute to Qutest.

seems odd...most reviews i have seen about the mojo considered even the lower end bifrost mb superior to the mojo...in all the reviews i have seen about the mojo in comparison to the schiit suggested it is more "rounded" and i read 3 different reviews that said the same.  I have been considering both the chord and schiit for awhile and was leaning more toward the chords for the dsd support, but after reading and "believing" i better understand the purpose and advantage of upsampling, i am now leaning more toward schiit.  I would have thought you or anyone would have been fairly impressed with gungnir multibit with unison....how much are you going to want for it...i might be a buyer for it?!@

Is it a newer one with the new analog board or did  you just have an older one upgraded?

[beerandmusic]

1 hour ago, davide256 said:

Qobuz. looses some cymbal and low base definition compared to when I compare same recordings on NAS but still very good. An iPhone and Qobuz makes

DAP's seem cost ineffective.

I don't do apple, but I was thinking the same but haven't yet tried android as a DAP connected to a dac, although even connecting an android directly to an amp sounds relatively impressive.

 

I am curious, do you or anyone know if the newer android phones with type c connector (i have a moto g7), allow you to connect to a usb dac?

[beerandmusic]

Back to the original question, i believe the guy that connects a noisy laptop directly to a 25K meitner does not lose anything compared to those that stream...I even think that any decent dac that uses their own clean 5v reference voltage, is galvanically isolated, and reclocks the signal should compensate for noise.

 

I believe those that stream compensate for noisy usb in different ways and that both methods are effective if done right.

 

I also believe that the majority of those that stream are into upsampling where noise can be a bigger issue, but you really can't better SQ than upsampling to say 192K pcm.

 

44.1K fully covers our entire hearing capability, and upsampling is just to push distortion above the audio spectrum during analog conversion, and pcm 192k will introduce less jitter than dsd rates and be easier to manage and anything over that is overkill, and just creating more issues to resolve for....keep it simple.

 

There will never be more audible music in high rate dsd compared to a clean 192K pcm signal, which is much easier to keep clean.

[beerandmusic]

9 minutes ago, The Computer Audiophile said:

Can you explain this one?

 

which part....

a) no audible music above 192k

b) 192K easier to keep clean

 

i am guessing part b since most people acknowledge that 44.1K encompasses the audible spectrum.

inre b, i am not saying that higher upsampling rates can't be done with neglible audible differences, but it does take extra work that is unnecessary.

 

[beerandmusic]

2 hours ago, davide256 said:

in transition. selling Gungnir, mojo is interim solution enroute to Qutest.

 

I re-read and i guess the analog 2 is only for the yggi....how much do you want for your gungnir mb w/unison? And is it a newer gungnir that came with unison or an older one you had upgraded?

[beerandmusic]

12 minutes ago, The Computer Audiophile said:

B. 

i just edited answer above, and as you know I just echo others input... i am not a dac engineer, but from what I understand, the only purpose of upsampling is::

 

to quote PKANE::

The point of upsampling is to move the reconstruction/anti-aliasing filter well above audible frequencies, where it can be more gentle and not affect the audible spectrum. A high-quality filter at 22khz is hard to make in hardware without distorting phase and without cutting into frequencies below 20khz. A gentle filter at 88khz is much easier, and it can distort there all it wants without affecting the audible range. It's a simple engineering solution to a problem that could otherwise become audible... at least to some of us (not to me, not for a while )

to quote MONTY MONTGOMERY::

Monty Montgomery https://www.youtube.com/watch?v=cIQ9IXSUzuM&feature=youtu.be

If the ADC/DAC obeys the Nyquist rules (bandlimited) then your waveform will be perfectly reconstructed without additional interpolation. As mentioned its about making filtering easier, but that has become a bit of a moot point with current DACs.

----

these i would have to look up reliable sources, but i "believe" is universally accepted

Upsampling requires processing power and power which can introduce it's own noise if you have don't have quality power supplies.

All upsampling adds artifacts and changes the signal that has to be compensated for.

Is there any logical reasoning to upsample higher than 192K?

I will find reliable sources stating such if there is any problem you have with anything in this post?

I "believe" Miska would be best to answer what upsampling past 192K PCM actually improves on, and what are potential issues?  But without a logical explanation from him, that I am free to debate, I would have a hard time accepting.

 

[beerandmusic]

8 minutes ago, The Computer Audiophile said:

I just don’t see how any of that explains your comment here:

 

 

Perhaps you should ask @Miska, who actually designs filters, about this stuff. 

 

I edited post above and added MISKA would be the best to answer before you did (smile).  I agree, but even you answered it yourself...you have to design filters to clean up the mess....why create the mess in the first place, when it doesn't give you any more than 192K PCM...again, he could probably answer, and if he was honest, he would likely say that it is subjective (at best) whether it sounds better than 192K PCM without the additional upsampling and filtering.

[beerandmusic]

8 minutes ago, The Computer Audiophile said:

Oh boy. I kindly suggest you hold up on offering any more "information" about filters etc... It isn't in your wheelhouse. 

that's why i said to ask MISKA...i didn't say anything about filters...I know they are used for the reconstruction, that is all i know, and won't pretend i understand more.

I only suggest the higher rate DSD will require filters unneeded for lower rate PCM...both require filtering, but I "believe" even MISKA will suggest more advanced filtering is needed for higher rate DSD than lower rate PCM.

[beerandmusic]

37 minutes ago, The Computer Audiophile said:

 

TO quote mojo-audio::

 

DSD has significantly higher quantization noise than PCM, and the noise is much closer to audible frequencies, requiring significantly more sophisticated digital filters, as well as noise-shaping and upsampling algorithms. The algorithms native DSD DACs use often result in an overly smoothed over sound without the same immediacy, articulation, and harmonic coherency R-2R ladder DACs are known for.

 

Granted he does suggest using hqplayer to do quad dsd if you are going to do DSD at all, but he also suggest playing music in it's native format.

 

Again, why go from 44.1K PCM to quad rate DSD and all the expense and process power by modifying and filtering just to play something that was recorded at 44.1K....when that rate encompasses our hearing spectrum.

 

He also states::

The truth is that in recent blind studies they've proved that high-resolution PCM and DSD are statistically indistinguishable from one another.

[beerandmusic]

23 minutes ago, The Computer Audiophile said:

i would never quote Mojo Audio on anything. 

haha....maybe MISKA will step in and correct him...then i would believe it...i trust everything MISKA says (smile)...imho, Miska is the most respected person on this site, with Jabbr a close second.

 

I believe i remember at one point that miska while debating dsd over pcm suggested that one of the main benefits is you can use a cheaper dac....correct me if I am wrong?

 

Miska, if you are there, is there any benefit of Quad rate dsd over 192K PCM if the original recording is 44.1K PCM?  (assume same hardware or if it makes more sense consider all same hardware except dac e.g. schiit gungnir mb w/unison vs RME)